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-rw-r--r--src/output/alsa_plugin.c444
-rw-r--r--src/output/ao_plugin.c253
-rw-r--r--src/output/fifo_plugin.c290
-rw-r--r--src/output/jack_plugin.c486
-rw-r--r--src/output/mvp_plugin.c280
-rw-r--r--src/output/null_plugin.c85
-rw-r--r--src/output/oss_plugin.c571
-rw-r--r--src/output/osx_plugin.c368
-rw-r--r--src/output/pulse_plugin.c218
-rw-r--r--src/output/shout_mp3.c188
-rw-r--r--src/output/shout_ogg.c306
-rw-r--r--src/output/shout_plugin.c596
-rw-r--r--src/output/shout_plugin.h93
13 files changed, 4178 insertions, 0 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
new file mode 100644
index 000000000..1845f1b76
--- /dev/null
+++ b/src/output/alsa_plugin.c
@@ -0,0 +1,444 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+
+#ifdef HAVE_ALSA
+
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+
+static const char default_device[] = "default";
+
+#define MPD_ALSA_RETRY_NR 5
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <alsa/asoundlib.h>
+
+typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
+ snd_pcm_uframes_t size);
+
+typedef struct _AlsaData {
+ const char *device;
+
+ /** the mode flags passed to snd_pcm_open */
+ int mode;
+
+ snd_pcm_t *pcmHandle;
+ alsa_writei_t *writei;
+ unsigned int buffer_time;
+ unsigned int period_time;
+ int sampleSize;
+ int useMmap;
+} AlsaData;
+
+static AlsaData *newAlsaData(void)
+{
+ AlsaData *ret = xmalloc(sizeof(AlsaData));
+
+ ret->device = default_device;
+ ret->mode = 0;
+ ret->pcmHandle = NULL;
+ ret->writei = snd_pcm_writei;
+ ret->useMmap = 0;
+ ret->buffer_time = 0;
+ ret->period_time = 0;
+
+ return ret;
+}
+
+static void freeAlsaData(AlsaData * ad)
+{
+ if (ad->device && ad->device != default_device)
+ xfree(ad->device);
+ free(ad);
+}
+
+static void
+alsa_configure(AlsaData *ad, ConfigParam *param)
+{
+ BlockParam *bp;
+
+ if ((bp = getBlockParam(param, "device")))
+ ad->device = xstrdup(bp->value);
+ ad->useMmap = getBoolBlockParam(param, "use_mmap", 1);
+ if (ad->useMmap == CONF_BOOL_UNSET)
+ ad->useMmap = 0;
+ if ((bp = getBlockParam(param, "buffer_time")))
+ ad->buffer_time = atoi(bp->value);
+ if ((bp = getBlockParam(param, "period_time")))
+ ad->period_time = atoi(bp->value);
+
+#ifdef SND_PCM_NO_AUTO_RESAMPLE
+ if (!getBoolBlockParam(param, "auto_resample", true))
+ ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
+#endif
+
+#ifdef SND_PCM_NO_AUTO_CHANNELS
+ if (!getBoolBlockParam(param, "auto_channels", true))
+ ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
+#endif
+
+#ifdef SND_PCM_NO_AUTO_FORMAT
+ if (!getBoolBlockParam(param, "auto_format", true))
+ ad->mode |= SND_PCM_NO_AUTO_FORMAT;
+#endif
+}
+
+static void *alsa_initDriver(mpd_unused struct audio_output *ao,
+ mpd_unused const struct audio_format *audio_format,
+ ConfigParam * param)
+{
+ /* no need for pthread_once thread-safety when reading config */
+ static int free_global_registered;
+ AlsaData *ad = newAlsaData();
+
+ if (!free_global_registered) {
+ atexit((void(*)(void))snd_config_update_free_global);
+ free_global_registered = 1;
+ }
+
+ if (param)
+ alsa_configure(ad, param);
+
+ return ad;
+}
+
+static void alsa_finishDriver(void *data)
+{
+ AlsaData *ad = data;
+
+ freeAlsaData(ad);
+}
+
+static int alsa_testDefault(void)
+{
+ snd_pcm_t *handle;
+
+ int ret = snd_pcm_open(&handle, default_device,
+ SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+ if (ret) {
+ WARNING("Error opening default ALSA device: %s\n",
+ snd_strerror(-ret));
+ return -1;
+ } else
+ snd_pcm_close(handle);
+
+ return 0;
+}
+
+static snd_pcm_format_t get_bitformat(const struct audio_format *af)
+{
+ switch (af->bits) {
+ case 8: return SND_PCM_FORMAT_S8;
+ case 16: return SND_PCM_FORMAT_S16;
+ case 24: return SND_PCM_FORMAT_S24;
+ case 32: return SND_PCM_FORMAT_S32;
+ }
+ return SND_PCM_FORMAT_UNKNOWN;
+}
+
+static int alsa_openDevice(void *data, struct audio_format *audioFormat)
+{
+ AlsaData *ad = data;
+ snd_pcm_format_t bitformat;
+ snd_pcm_hw_params_t *hwparams;
+ snd_pcm_sw_params_t *swparams;
+ unsigned int sample_rate = audioFormat->sample_rate;
+ unsigned int channels = audioFormat->channels;
+ snd_pcm_uframes_t alsa_buffer_size;
+ snd_pcm_uframes_t alsa_period_size;
+ int err;
+ const char *cmd = NULL;
+ int retry = MPD_ALSA_RETRY_NR;
+ unsigned int period_time, period_time_ro;
+ unsigned int buffer_time;
+
+ if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN)
+ ERROR("ALSA device \"%s\" doesn't support %u bit audio\n",
+ ad->device, audioFormat->bits);
+
+ err = snd_pcm_open(&ad->pcmHandle, ad->device,
+ SND_PCM_STREAM_PLAYBACK, ad->mode);
+ if (err < 0) {
+ ad->pcmHandle = NULL;
+ goto error;
+ }
+
+ period_time_ro = period_time = ad->period_time;
+configure_hw:
+ /* configure HW params */
+ snd_pcm_hw_params_alloca(&hwparams);
+
+ cmd = "snd_pcm_hw_params_any";
+ err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
+ if (err < 0)
+ goto error;
+
+ if (ad->useMmap) {
+ err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
+ SND_PCM_ACCESS_MMAP_INTERLEAVED);
+ if (err < 0) {
+ ERROR("Cannot set mmap'ed mode on ALSA device \"%s\": "
+ " %s\n", ad->device, snd_strerror(-err));
+ ERROR("Falling back to direct write mode\n");
+ ad->useMmap = 0;
+ } else
+ ad->writei = snd_pcm_mmap_writei;
+ }
+
+ if (!ad->useMmap) {
+ cmd = "snd_pcm_hw_params_set_access";
+ err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (err < 0)
+ goto error;
+ ad->writei = snd_pcm_writei;
+ }
+
+ err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
+ if (err == -EINVAL && audioFormat->bits != 16) {
+ /* fall back to 16 bit, let pcm_utils.c do the conversion */
+ err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams,
+ SND_PCM_FORMAT_S16);
+ if (err == 0) {
+ DEBUG("ALSA device \"%s\": converting %u bit to 16 bit\n",
+ ad->device, audioFormat->bits);
+ audioFormat->bits = 16;
+ }
+ }
+
+ if (err < 0) {
+ ERROR("ALSA device \"%s\" does not support %u bit audio: "
+ "%s\n", ad->device, audioFormat->bits, snd_strerror(-err));
+ goto fail;
+ }
+
+ err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
+ &channels);
+ if (err < 0) {
+ ERROR("ALSA device \"%s\" does not support %i channels: "
+ "%s\n", ad->device, (int)audioFormat->channels,
+ snd_strerror(-err));
+ goto fail;
+ }
+ audioFormat->channels = (int8_t)channels;
+
+ err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
+ &sample_rate, NULL);
+ if (err < 0 || sample_rate == 0) {
+ ERROR("ALSA device \"%s\" does not support %u Hz audio\n",
+ ad->device, audioFormat->sample_rate);
+ goto fail;
+ }
+ audioFormat->sample_rate = sample_rate;
+
+ if (ad->buffer_time > 0) {
+ buffer_time = ad->buffer_time;
+ cmd = "snd_pcm_hw_params_set_buffer_time_near";
+ err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
+ &buffer_time, NULL);
+ if (err < 0)
+ goto error;
+ }
+
+ if (period_time_ro > 0) {
+ period_time = period_time_ro;
+ cmd = "snd_pcm_hw_params_set_period_time_near";
+ err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
+ &period_time, NULL);
+ if (err < 0)
+ goto error;
+ }
+
+ cmd = "snd_pcm_hw_params";
+ err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
+ if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
+ period_time_ro = period_time_ro >> 1;
+ goto configure_hw;
+ } else if (err < 0)
+ goto error;
+ if (retry != MPD_ALSA_RETRY_NR)
+ DEBUG("ALSA period_time set to %d\n", period_time);
+
+ cmd = "snd_pcm_hw_params_get_buffer_size";
+ err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_hw_params_get_period_size";
+ err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
+ NULL);
+ if (err < 0)
+ goto error;
+
+ /* configure SW params */
+ snd_pcm_sw_params_alloca(&swparams);
+
+ cmd = "snd_pcm_sw_params_current";
+ err = snd_pcm_sw_params_current(ad->pcmHandle, swparams);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params_set_start_threshold";
+ err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
+ alsa_buffer_size -
+ alsa_period_size);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params_set_avail_min";
+ err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams,
+ alsa_period_size);
+ if (err < 0)
+ goto error;
+
+ cmd = "snd_pcm_sw_params";
+ err = snd_pcm_sw_params(ad->pcmHandle, swparams);
+ if (err < 0)
+ goto error;
+
+ ad->sampleSize = audio_format_frame_size(audioFormat);
+
+ DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at "
+ "%u Hz\n", ad->device, audioFormat->bits,
+ channels, sample_rate);
+
+ return 0;
+
+error:
+ if (cmd) {
+ ERROR("Error opening ALSA device \"%s\" (%s): %s\n",
+ ad->device, cmd, snd_strerror(-err));
+ } else {
+ ERROR("Error opening ALSA device \"%s\": %s\n", ad->device,
+ snd_strerror(-err));
+ }
+fail:
+ if (ad->pcmHandle)
+ snd_pcm_close(ad->pcmHandle);
+ ad->pcmHandle = NULL;
+ return -1;
+}
+
+static int alsa_errorRecovery(AlsaData * ad, int err)
+{
+ if (err == -EPIPE) {
+ DEBUG("Underrun on ALSA device \"%s\"\n", ad->device);
+ } else if (err == -ESTRPIPE) {
+ DEBUG("ALSA device \"%s\" was suspended\n", ad->device);
+ }
+
+ switch (snd_pcm_state(ad->pcmHandle)) {
+ case SND_PCM_STATE_PAUSED:
+ err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
+ break;
+ case SND_PCM_STATE_SUSPENDED:
+ err = snd_pcm_resume(ad->pcmHandle);
+ if (err == -EAGAIN)
+ return 0;
+ /* fall-through to snd_pcm_prepare: */
+ case SND_PCM_STATE_SETUP:
+ case SND_PCM_STATE_XRUN:
+ err = snd_pcm_prepare(ad->pcmHandle);
+ break;
+ case SND_PCM_STATE_DISCONNECTED:
+ /* so alsa_closeDevice won't try to drain: */
+ snd_pcm_close(ad->pcmHandle);
+ ad->pcmHandle = NULL;
+ break;
+ /* this is no error, so just keep running */
+ case SND_PCM_STATE_RUNNING:
+ err = 0;
+ break;
+ default:
+ /* unknown state, do nothing */
+ break;
+ }
+
+ return err;
+}
+
+static void alsa_dropBufferedAudio(void *data)
+{
+ AlsaData *ad = data;
+
+ alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
+}
+
+static void alsa_closeDevice(void *data)
+{
+ AlsaData *ad = data;
+
+ if (ad->pcmHandle) {
+ if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) {
+ snd_pcm_drain(ad->pcmHandle);
+ }
+ snd_pcm_close(ad->pcmHandle);
+ ad->pcmHandle = NULL;
+ }
+}
+
+static int alsa_playAudio(void *data, const char *playChunk, size_t size)
+{
+ AlsaData *ad = data;
+ int ret;
+
+ size /= ad->sampleSize;
+
+ while (size > 0) {
+ ret = ad->writei(ad->pcmHandle, playChunk, size);
+
+ if (ret == -EAGAIN || ret == -EINTR)
+ continue;
+
+ if (ret < 0) {
+ if (alsa_errorRecovery(ad, ret) < 0) {
+ ERROR("closing ALSA device \"%s\" due to write "
+ "error: %s\n", ad->device,
+ snd_strerror(-errno));
+ alsa_closeDevice(ad);
+ return -1;
+ }
+ continue;
+ }
+
+ playChunk += ret * ad->sampleSize;
+ size -= ret;
+ }
+
+ return 0;
+}
+
+const struct audio_output_plugin alsaPlugin = {
+ .name = "alsa",
+ .test_default_device = alsa_testDefault,
+ .init = alsa_initDriver,
+ .finish = alsa_finishDriver,
+ .open = alsa_openDevice,
+ .play = alsa_playAudio,
+ .cancel = alsa_dropBufferedAudio,
+ .close = alsa_closeDevice,
+};
+
+#else /* HAVE ALSA */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
+#endif /* HAVE_ALSA */
diff --git a/src/output/ao_plugin.c b/src/output/ao_plugin.c
new file mode 100644
index 000000000..e731f972a
--- /dev/null
+++ b/src/output/ao_plugin.c
@@ -0,0 +1,253 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+
+#ifdef HAVE_AO
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <ao/ao.h>
+
+static int driverInitCount;
+
+typedef struct _AoData {
+ int writeSize;
+ int driverId;
+ ao_option *options;
+ ao_device *device;
+} AoData;
+
+static AoData *newAoData(void)
+{
+ AoData *ret = xmalloc(sizeof(AoData));
+ ret->device = NULL;
+ ret->options = NULL;
+
+ return ret;
+}
+
+static void audioOutputAo_error(void)
+{
+ if (errno == AO_ENOTLIVE) {
+ ERROR("not a live ao device\n");
+ } else if (errno == AO_EOPENDEVICE) {
+ ERROR("not able to open audio device\n");
+ } else if (errno == AO_EBADOPTION) {
+ ERROR("bad driver option\n");
+ }
+}
+
+static void *audioOutputAo_initDriver(struct audio_output *ao,
+ mpd_unused const struct audio_format *audio_format,
+ ConfigParam * param)
+{
+ ao_info *ai;
+ char *duplicated;
+ char *stk1;
+ char *stk2;
+ char *n1;
+ char *key;
+ char *value;
+ char *test;
+ AoData *ad = newAoData();
+ BlockParam *blockParam;
+
+ if ((blockParam = getBlockParam(param, "write_size"))) {
+ ad->writeSize = strtol(blockParam->value, &test, 10);
+ if (*test != '\0') {
+ FATAL("\"%s\" is not a valid write size at line %i\n",
+ blockParam->value, blockParam->line);
+ }
+ } else
+ ad->writeSize = 1024;
+
+ if (driverInitCount == 0) {
+ ao_initialize();
+ }
+ driverInitCount++;
+
+ blockParam = getBlockParam(param, "driver");
+
+ if (!blockParam || 0 == strcmp(blockParam->value, "default")) {
+ ad->driverId = ao_default_driver_id();
+ } else if ((ad->driverId = ao_driver_id(blockParam->value)) < 0) {
+ FATAL("\"%s\" is not a valid ao driver at line %i\n",
+ blockParam->value, blockParam->line);
+ }
+
+ if ((ai = ao_driver_info(ad->driverId)) == NULL) {
+ FATAL("problems getting driver info for device defined at line %i\n"
+ "you may not have permission to the audio device\n", param->line);
+ }
+
+ DEBUG("using ao driver \"%s\" for \"%s\"\n", ai->short_name,
+ audio_output_get_name(ao));
+
+ blockParam = getBlockParam(param, "options");
+
+ if (blockParam) {
+ duplicated = xstrdup(blockParam->value);
+ } else
+ duplicated = xstrdup("");
+
+ if (strlen(duplicated)) {
+ stk1 = NULL;
+ n1 = strtok_r(duplicated, ";", &stk1);
+ while (n1) {
+ stk2 = NULL;
+ key = strtok_r(n1, "=", &stk2);
+ if (!key)
+ FATAL("problems parsing options \"%s\"\n", n1);
+ /*found = 0;
+ for(i=0;i<ai->option_count;i++) {
+ if(strcmp(ai->options[i],key)==0) {
+ found = 1;
+ break;
+ }
+ }
+ if(!found) {
+ FATAL("\"%s\" is not an option for "
+ "\"%s\" ao driver\n",key,
+ ai->short_name);
+ } */
+ value = strtok_r(NULL, "", &stk2);
+ if (!value)
+ FATAL("problems parsing options \"%s\"\n", n1);
+ ao_append_option(&ad->options, key, value);
+ n1 = strtok_r(NULL, ";", &stk1);
+ }
+ }
+ free(duplicated);
+
+ return ad;
+}
+
+static void freeAoData(AoData * ad)
+{
+ ao_free_options(ad->options);
+ free(ad);
+}
+
+static void audioOutputAo_finishDriver(void *data)
+{
+ AoData *ad = (AoData *)data;
+ freeAoData(ad);
+
+ driverInitCount--;
+
+ if (driverInitCount == 0)
+ ao_shutdown();
+}
+
+static void audioOutputAo_dropBufferedAudio(mpd_unused void *data)
+{
+ /* not supported by libao */
+}
+
+static void audioOutputAo_closeDevice(void *data)
+{
+ AoData *ad = (AoData *)data;
+
+ if (ad->device) {
+ ao_close(ad->device);
+ ad->device = NULL;
+ }
+}
+
+static int audioOutputAo_openDevice(void *data,
+ struct audio_format *audio_format)
+{
+ ao_sample_format format;
+ AoData *ad = (AoData *)data;
+
+ if (ad->device) {
+ audioOutputAo_closeDevice(ad);
+ }
+
+ format.bits = audio_format->bits;
+ format.rate = audio_format->sample_rate;
+ format.byte_format = AO_FMT_NATIVE;
+ format.channels = audio_format->channels;
+
+ ad->device = ao_open_live(ad->driverId, &format, ad->options);
+
+ if (ad->device == NULL)
+ return -1;
+
+ return 0;
+}
+
+/**
+ * For whatever reason, libao wants a non-const pointer. Let's hope
+ * it does not write to the buffer, and use the union deconst hack to
+ * work around this API misdesign.
+ */
+static int ao_play_deconst(ao_device *device, const void *output_samples,
+ uint_32 num_bytes)
+{
+ union {
+ const void *in;
+ void *out;
+ } u;
+
+ u.in = output_samples;
+ return ao_play(device, u.out, num_bytes);
+}
+
+static int audioOutputAo_play(void *data, const char *playChunk, size_t size)
+{
+ AoData *ad = (AoData *)data;
+ size_t chunk_size;
+
+ if (ad->device == NULL)
+ return -1;
+
+ while (size > 0) {
+ chunk_size = (size_t)ad->writeSize > size
+ ? size : (size_t)ad->writeSize;
+
+ if (ao_play_deconst(ad->device, playChunk, chunk_size) == 0) {
+ audioOutputAo_error();
+ ERROR("closing audio device due to write error\n");
+ audioOutputAo_closeDevice(ad);
+ return -1;
+ }
+
+ playChunk += chunk_size;
+ size -= chunk_size;
+ }
+
+ return 0;
+}
+
+const struct audio_output_plugin aoPlugin = {
+ .name = "ao",
+ .init = audioOutputAo_initDriver,
+ .finish = audioOutputAo_finishDriver,
+ .open = audioOutputAo_openDevice,
+ .play = audioOutputAo_play,
+ .cancel = audioOutputAo_dropBufferedAudio,
+ .close = audioOutputAo_closeDevice,
+};
+
+#else
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(aoPlugin)
+#endif
diff --git a/src/output/fifo_plugin.c b/src/output/fifo_plugin.c
new file mode 100644
index 000000000..d7eb91ff6
--- /dev/null
+++ b/src/output/fifo_plugin.c
@@ -0,0 +1,290 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+
+#ifdef HAVE_FIFO
+
+#include "../log.h"
+#include "../utils.h"
+#include "../timer.h"
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
+#define FIFO_BUFFER_SIZE 65536 /* pipe capacity on Linux >= 2.6.11 */
+
+typedef struct _FifoData {
+ char *path;
+ int input;
+ int output;
+ int created;
+ Timer *timer;
+} FifoData;
+
+static FifoData *newFifoData(void)
+{
+ FifoData *ret;
+
+ ret = xmalloc(sizeof(FifoData));
+
+ ret->path = NULL;
+ ret->input = -1;
+ ret->output = -1;
+ ret->created = 0;
+ ret->timer = NULL;
+
+ return ret;
+}
+
+static void freeFifoData(FifoData *fd)
+{
+ if (fd->path)
+ free(fd->path);
+
+ if (fd->timer)
+ timer_free(fd->timer);
+
+ free(fd);
+}
+
+static void removeFifo(FifoData *fd)
+{
+ DEBUG("Removing FIFO \"%s\"\n", fd->path);
+
+ if (unlink(fd->path) < 0) {
+ ERROR("Could not remove FIFO \"%s\": %s\n",
+ fd->path, strerror(errno));
+ return;
+ }
+
+ fd->created = 0;
+}
+
+static void closeFifo(FifoData *fd)
+{
+ struct stat st;
+
+ if (fd->input >= 0) {
+ close(fd->input);
+ fd->input = -1;
+ }
+
+ if (fd->output >= 0) {
+ close(fd->output);
+ fd->output = -1;
+ }
+
+ if (fd->created && (stat(fd->path, &st) == 0))
+ removeFifo(fd);
+}
+
+static int makeFifo(FifoData *fd)
+{
+ if (mkfifo(fd->path, 0666) < 0) {
+ ERROR("Couldn't create FIFO \"%s\": %s\n",
+ fd->path, strerror(errno));
+ return -1;
+ }
+
+ fd->created = 1;
+
+ return 0;
+}
+
+static int checkFifo(FifoData *fd)
+{
+ struct stat st;
+
+ if (stat(fd->path, &st) < 0) {
+ if (errno == ENOENT) {
+ /* Path doesn't exist */
+ return makeFifo(fd);
+ }
+
+ ERROR("Failed to stat FIFO \"%s\": %s\n",
+ fd->path, strerror(errno));
+ return -1;
+ }
+
+ if (!S_ISFIFO(st.st_mode)) {
+ ERROR("\"%s\" already exists, but is not a FIFO\n", fd->path);
+ return -1;
+ }
+
+ return 0;
+}
+
+static int openFifo(FifoData *fd)
+{
+ if (checkFifo(fd) < 0)
+ return -1;
+
+ fd->input = open(fd->path, O_RDONLY|O_NONBLOCK);
+ if (fd->input < 0) {
+ ERROR("Could not open FIFO \"%s\" for reading: %s\n",
+ fd->path, strerror(errno));
+ closeFifo(fd);
+ return -1;
+ }
+
+ fd->output = open(fd->path, O_WRONLY|O_NONBLOCK);
+ if (fd->output < 0) {
+ ERROR("Could not open FIFO \"%s\" for writing: %s\n",
+ fd->path, strerror(errno));
+ closeFifo(fd);
+ return -1;
+ }
+
+ return 0;
+}
+
+static void *fifo_initDriver(mpd_unused struct audio_output *ao,
+ mpd_unused const struct audio_format *audio_format,
+ ConfigParam *param)
+{
+ FifoData *fd;
+ BlockParam *blockParam;
+ char *path;
+
+ blockParam = getBlockParam(param, "path");
+ if (!blockParam) {
+ FATAL("No \"path\" parameter specified for fifo output "
+ "defined at line %i\n", param->line);
+ }
+
+ path = parsePath(blockParam->value);
+ if (!path) {
+ FATAL("Could not parse \"path\" parameter for fifo output "
+ "at line %i\n", blockParam->line);
+ }
+
+ fd = newFifoData();
+ fd->path = path;
+
+ if (openFifo(fd) < 0) {
+ freeFifoData(fd);
+ return NULL;
+ }
+
+ return fd;
+}
+
+static void fifo_finishDriver(void *data)
+{
+ FifoData *fd = (FifoData *)data;
+
+ closeFifo(fd);
+ freeFifoData(fd);
+}
+
+static int fifo_openDevice(void *data,
+ struct audio_format *audio_format)
+{
+ FifoData *fd = (FifoData *)data;
+
+ if (fd->timer)
+ timer_free(fd->timer);
+
+ fd->timer = timer_new(audio_format);
+
+ return 0;
+}
+
+static void fifo_closeDevice(void *data)
+{
+ FifoData *fd = (FifoData *)data;
+
+ if (fd->timer) {
+ timer_free(fd->timer);
+ fd->timer = NULL;
+ }
+}
+
+static void fifo_dropBufferedAudio(void *data)
+{
+ FifoData *fd = (FifoData *)data;
+ char buf[FIFO_BUFFER_SIZE];
+ int bytes = 1;
+
+ timer_reset(fd->timer);
+
+ while (bytes > 0 && errno != EINTR)
+ bytes = read(fd->input, buf, FIFO_BUFFER_SIZE);
+
+ if (bytes < 0 && errno != EAGAIN) {
+ WARNING("Flush of FIFO \"%s\" failed: %s\n",
+ fd->path, strerror(errno));
+ }
+}
+
+static int fifo_playAudio(void *data,
+ const char *playChunk, size_t size)
+{
+ FifoData *fd = (FifoData *)data;
+ size_t offset = 0;
+ ssize_t bytes;
+
+ if (!fd->timer->started)
+ timer_start(fd->timer);
+ else
+ timer_sync(fd->timer);
+
+ timer_add(fd->timer, size);
+
+ while (size) {
+ bytes = write(fd->output, playChunk + offset, size);
+ if (bytes < 0) {
+ switch (errno) {
+ case EAGAIN:
+ /* The pipe is full, so empty it */
+ fifo_dropBufferedAudio(fd);
+ continue;
+ case EINTR:
+ continue;
+ }
+
+ ERROR("Closing FIFO output \"%s\" due to write error: "
+ "%s\n", fd->path, strerror(errno));
+ fifo_closeDevice(fd);
+ return -1;
+ }
+
+ size -= bytes;
+ offset += bytes;
+ }
+
+ return 0;
+}
+
+const struct audio_output_plugin fifoPlugin = {
+ .name = "fifo",
+ .init = fifo_initDriver,
+ .finish = fifo_finishDriver,
+ .open = fifo_openDevice,
+ .play = fifo_playAudio,
+ .cancel = fifo_dropBufferedAudio,
+ .close = fifo_closeDevice,
+};
+
+#else /* HAVE_FIFO */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(fifoPlugin)
+
+#endif /* !HAVE_FIFO */
diff --git a/src/output/jack_plugin.c b/src/output/jack_plugin.c
new file mode 100644
index 000000000..6f2bcd3a1
--- /dev/null
+++ b/src/output/jack_plugin.c
@@ -0,0 +1,486 @@
+/* jack plug in for the Music Player Daemon (MPD)
+ * (c)2006 by anarch(anarchsss@gmail.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+
+#ifdef HAVE_JACK
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <assert.h>
+
+#include <jack/jack.h>
+#include <jack/types.h>
+#include <jack/ringbuffer.h>
+
+static const size_t sample_size = sizeof(jack_default_audio_sample_t);
+
+struct jack_data {
+ struct audio_output *ao;
+
+ /* configuration */
+ const char *name;
+ const char *output_ports[2];
+ int ringbuffer_size;
+
+ /* for srate() only */
+ struct audio_format *audio_format;
+
+ /* jack library stuff */
+ jack_port_t *ports[2];
+ jack_client_t *client;
+ jack_ringbuffer_t *ringbuffer[2];
+ int bps;
+ int shutdown;
+};
+
+static struct jack_data *
+mpd_jack_new(void)
+{
+ struct jack_data *ret;
+
+ ret = xcalloc(sizeof(*ret), 1);
+
+ ret->name = "mpd";
+ ret->ringbuffer_size = 32768;
+
+ return ret;
+}
+
+static void
+mpd_jack_client_free(struct jack_data *jd)
+{
+ assert(jd != NULL);
+
+ if (jd->client != NULL) {
+ jack_deactivate(jd->client);
+ jack_client_close(jd->client);
+ jd->client = NULL;
+ }
+
+ if (jd->ringbuffer[0] != NULL) {
+ jack_ringbuffer_free(jd->ringbuffer[0]);
+ jd->ringbuffer[0] = NULL;
+ }
+
+ if (jd->ringbuffer[1] != NULL) {
+ jack_ringbuffer_free(jd->ringbuffer[1]);
+ jd->ringbuffer[1] = NULL;
+ }
+}
+
+static void
+mpd_jack_free(struct jack_data *jd)
+{
+ int i;
+
+ assert(jd != NULL);
+
+ mpd_jack_client_free(jd);
+
+ if (strcmp(jd->name, "mpd") != 0)
+ xfree(jd->name);
+
+ for ( i = ARRAY_SIZE(jd->output_ports); --i >= 0; ) {
+ if (!jd->output_ports[i])
+ continue;
+ xfree(jd->output_ports[i]);
+ }
+
+ free(jd);
+}
+
+static void
+mpd_jack_finish(void *data)
+{
+ struct jack_data *jd = data;
+ mpd_jack_free(jd);
+}
+
+static int
+mpd_jack_srate(mpd_unused jack_nframes_t rate, void *data)
+{
+ struct jack_data *jd = (struct jack_data *)data;
+ struct audio_format *audioFormat = jd->audio_format;
+
+ audioFormat->sample_rate = (int)jack_get_sample_rate(jd->client);
+
+ return 0;
+}
+
+static int
+mpd_jack_process(jack_nframes_t nframes, void *arg)
+{
+ struct jack_data *jd = (struct jack_data *) arg;
+ jack_default_audio_sample_t *out;
+ size_t available;
+
+ if (nframes <= 0)
+ return 0;
+
+ for (unsigned i = 0; i < 2; ++i) {
+ available = jack_ringbuffer_read_space(jd->ringbuffer[i]);
+ assert(available % sample_size == 0);
+ available /= sample_size;
+ if (available > nframes)
+ available = nframes;
+
+ out = jack_port_get_buffer(jd->ports[i], nframes);
+ jack_ringbuffer_read(jd->ringbuffer[i],
+ (char *)out, available * sample_size);
+
+ while (available < nframes)
+ /* ringbuffer underrun, fill with silence */
+ out[available++] = 0.0;
+ }
+
+ return 0;
+}
+
+static void
+mpd_jack_shutdown(void *arg)
+{
+ struct jack_data *jd = (struct jack_data *) arg;
+ jd->shutdown = 1;
+}
+
+static void
+set_audioformat(struct jack_data *jd, struct audio_format *audio_format)
+{
+ audio_format->sample_rate = jack_get_sample_rate(jd->client);
+ DEBUG("samplerate = %u\n", audio_format->sample_rate);
+ audio_format->channels = 2;
+
+ if (audio_format->bits != 16 && audio_format->bits != 24)
+ audio_format->bits = 24;
+
+ jd->bps = audio_format->channels
+ * sizeof(jack_default_audio_sample_t)
+ * audio_format->sample_rate;
+}
+
+static void
+mpd_jack_error(const char *msg)
+{
+ ERROR("jack: %s\n", msg);
+}
+
+static void *
+mpd_jack_init(struct audio_output *ao,
+ mpd_unused const struct audio_format *audio_format,
+ ConfigParam *param)
+{
+ struct jack_data *jd;
+ BlockParam *bp;
+ char *endptr;
+ int val;
+ char *cp = NULL;
+
+ jd = mpd_jack_new();
+ jd->ao = ao;
+
+ DEBUG("mpd_jack_init (pid=%d)\n", getpid());
+ if (param == NULL)
+ return jd;
+
+ if ( (bp = getBlockParam(param, "ports")) ) {
+ DEBUG("output_ports=%s\n", bp->value);
+
+ if (!(cp = strchr(bp->value, ',')))
+ FATAL("expected comma and a second value for '%s' "
+ "at line %d: %s\n",
+ bp->name, bp->line, bp->value);
+
+ *cp = '\0';
+ jd->output_ports[0] = xstrdup(bp->value);
+ *cp++ = ',';
+
+ if (!*cp)
+ FATAL("expected a second value for '%s' at line %d: "
+ "%s\n", bp->name, bp->line, bp->value);
+
+ jd->output_ports[1] = xstrdup(cp);
+
+ if (strchr(cp,','))
+ FATAL("Only %d values are supported for '%s' "
+ "at line %d\n",
+ (int)ARRAY_SIZE(jd->output_ports),
+ bp->name, bp->line);
+ }
+
+ if ( (bp = getBlockParam(param, "ringbuffer_size")) ) {
+ errno = 0;
+ val = strtol(bp->value, &endptr, 10);
+
+ if ( errno == 0 && endptr != bp->value) {
+ jd->ringbuffer_size = val < 32768 ? 32768 : val;
+ DEBUG("ringbuffer_size=%d\n", jd->ringbuffer_size);
+ } else {
+ FATAL("%s is not a number; ringbuf_size=%d\n",
+ bp->value, jd->ringbuffer_size);
+ }
+ }
+
+ if ( (bp = getBlockParam(param, "name"))
+ && (strcmp(bp->value, "mpd") != 0) ) {
+ jd->name = xstrdup(bp->value);
+ DEBUG("name=%s\n", jd->name);
+ }
+
+ return jd;
+}
+
+static int
+mpd_jack_test_default_device(void)
+{
+ return 0;
+}
+
+static int
+mpd_jack_connect(struct jack_data *jd, struct audio_format *audio_format)
+{
+ const char **jports;
+ char *port_name;
+
+ jd->audio_format = audio_format;
+
+ if ( (jd->client = jack_client_new(jd->name)) == NULL ) {
+ ERROR("jack server not running?\n");
+ return -1;
+ }
+
+ jack_set_error_function(mpd_jack_error);
+ jack_set_process_callback(jd->client, mpd_jack_process, jd);
+ jack_set_sample_rate_callback(jd->client, mpd_jack_srate, jd);
+ jack_on_shutdown(jd->client, mpd_jack_shutdown, jd);
+
+ if ( jack_activate(jd->client) ) {
+ ERROR("cannot activate client\n");
+ return -1;
+ }
+
+ jd->ports[0] = jack_port_register(jd->client, "left",
+ JACK_DEFAULT_AUDIO_TYPE,
+ JackPortIsOutput, 0);
+ if ( !jd->ports[0] ) {
+ ERROR("Cannot register left output port.\n");
+ return -1;
+ }
+
+ jd->ports[1] = jack_port_register(jd->client, "right",
+ JACK_DEFAULT_AUDIO_TYPE,
+ JackPortIsOutput, 0);
+ if ( !jd->ports[1] ) {
+ ERROR("Cannot register right output port.\n");
+ return -1;
+ }
+
+ /* hay que buscar que hay */
+ if (!jd->output_ports[1] &&
+ (jports = jack_get_ports(jd->client, NULL, NULL,
+ JackPortIsPhysical | JackPortIsInput))) {
+ jd->output_ports[0] = jports[0];
+ jd->output_ports[1] = jports[1] ? jports[1] : jports[0];
+ DEBUG("output_ports: %s %s\n",
+ jd->output_ports[0], jd->output_ports[1]);
+ free(jports);
+ }
+
+ if ( jd->output_ports[1] ) {
+ jd->ringbuffer[0] = jack_ringbuffer_create(jd->ringbuffer_size);
+ jd->ringbuffer[1] = jack_ringbuffer_create(jd->ringbuffer_size);
+ memset(jd->ringbuffer[0]->buf, 0, jd->ringbuffer[0]->size);
+ memset(jd->ringbuffer[1]->buf, 0, jd->ringbuffer[1]->size);
+
+ port_name = xmalloc(sizeof(char)*(7+strlen(jd->name)));
+
+ sprintf(port_name, "%s:left", jd->name);
+ if ( (jack_connect(jd->client, port_name,
+ jd->output_ports[0])) != 0 ) {
+ ERROR("%s is not a valid Jack Client / Port\n",
+ jd->output_ports[0]);
+ free(port_name);
+ return -1;
+ }
+ sprintf(port_name, "%s:right", jd->name);
+ if ( (jack_connect(jd->client, port_name,
+ jd->output_ports[1])) != 0 ) {
+ ERROR("%s is not a valid Jack Client / Port\n",
+ jd->output_ports[1]);
+ free(port_name);
+ return -1;
+ }
+ free(port_name);
+ }
+
+ return 1;
+}
+
+static int
+mpd_jack_open(void *data, struct audio_format *audio_format)
+{
+ struct jack_data *jd = data;
+
+ assert(jd != NULL);
+
+ if (jd->client == NULL && mpd_jack_connect(jd, audio_format) < 0) {
+ mpd_jack_client_free(jd);
+ return -1;
+ }
+
+ set_audioformat(jd, audio_format);
+
+ return 0;
+}
+
+static void
+mpd_jack_close(mpd_unused void *data)
+{
+ /*mpd_jack_finish(audioOutput);*/
+}
+
+static void
+mpd_jack_cancel (mpd_unused void *data)
+{
+}
+
+static inline jack_default_audio_sample_t
+sample_16_to_jack(int16_t sample)
+{
+ return sample / (jack_default_audio_sample_t)(1 << (16 - 1));
+}
+
+static void
+mpd_jack_write_samples_16(struct jack_data *jd, const int16_t *src,
+ unsigned num_samples)
+{
+ jack_default_audio_sample_t sample;
+
+ while (num_samples-- > 0) {
+ sample = sample_16_to_jack(*src++);
+ jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
+ sizeof(sample));
+
+ sample = sample_16_to_jack(*src++);
+ jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
+ sizeof(sample));
+ }
+}
+
+static inline jack_default_audio_sample_t
+sample_24_to_jack(int32_t sample)
+{
+ return sample / (jack_default_audio_sample_t)(1 << (24 - 1));
+}
+
+static void
+mpd_jack_write_samples_24(struct jack_data *jd, const int32_t *src,
+ unsigned num_samples)
+{
+ jack_default_audio_sample_t sample;
+
+ while (num_samples-- > 0) {
+ sample = sample_24_to_jack(*src++);
+ jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
+ sizeof(sample));
+
+ sample = sample_24_to_jack(*src++);
+ jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
+ sizeof(sample));
+ }
+}
+
+static void
+mpd_jack_write_samples(struct jack_data *jd, const void *src,
+ unsigned num_samples)
+{
+ switch (jd->audio_format->bits) {
+ case 16:
+ mpd_jack_write_samples_16(jd, (const int16_t*)src,
+ num_samples);
+ break;
+
+ case 24:
+ mpd_jack_write_samples_24(jd, (const int32_t*)src,
+ num_samples);
+ break;
+
+ default:
+ assert(false);
+ }
+}
+
+static int
+mpd_jack_play(void *data, const char *buff, size_t size)
+{
+ struct jack_data *jd = data;
+ const size_t frame_size = audio_format_frame_size(jd->audio_format);
+ size_t space, space1;
+
+ if (jd->shutdown) {
+ ERROR("Refusing to play, because there is no client thread.\n");
+ mpd_jack_client_free(jd);
+ audio_output_closed(jd->ao);
+ return 0;
+ }
+
+ assert(size % frame_size == 0);
+ size /= frame_size;
+ while (size > 0 && !jd->shutdown) {
+ space = jack_ringbuffer_write_space(jd->ringbuffer[0]);
+ space1 = jack_ringbuffer_write_space(jd->ringbuffer[1]);
+ if (space > space1)
+ /* send data symmetrically */
+ space = space1;
+
+ space /= sample_size;
+ if (space > 0) {
+ if (space > size)
+ space = size;
+
+ mpd_jack_write_samples(jd, buff, space);
+
+ buff += space * frame_size;
+ size -= space;
+ } else {
+ /* XXX do something more intelligent to
+ synchronize */
+ my_usleep(10000);
+ }
+
+ }
+
+ return 0;
+}
+
+const struct audio_output_plugin jackPlugin = {
+ .name = "jack",
+ .test_default_device = mpd_jack_test_default_device,
+ .init = mpd_jack_init,
+ .finish = mpd_jack_finish,
+ .open = mpd_jack_open,
+ .play = mpd_jack_play,
+ .cancel = mpd_jack_cancel,
+ .close = mpd_jack_close,
+};
+
+#else /* HAVE JACK */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(jackPlugin)
+
+#endif /* HAVE_JACK */
diff --git a/src/output/mvp_plugin.c b/src/output/mvp_plugin.c
new file mode 100644
index 000000000..70dd25f9d
--- /dev/null
+++ b/src/output/mvp_plugin.c
@@ -0,0 +1,280 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * Media MVP audio output based on code from MVPMC project:
+ * http://mvpmc.sourceforge.net/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+
+#ifdef HAVE_MVP
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+
+typedef struct {
+ unsigned long dsp_status;
+ unsigned long stream_decode_type;
+ unsigned long sample_rate;
+ unsigned long bit_rate;
+ unsigned long raw[64 / sizeof(unsigned long)];
+} aud_status_t;
+
+#define MVP_SET_AUD_STOP _IOW('a',1,int)
+#define MVP_SET_AUD_PLAY _IOW('a',2,int)
+#define MVP_SET_AUD_PAUSE _IOW('a',3,int)
+#define MVP_SET_AUD_UNPAUSE _IOW('a',4,int)
+#define MVP_SET_AUD_SRC _IOW('a',5,int)
+#define MVP_SET_AUD_MUTE _IOW('a',6,int)
+#define MVP_SET_AUD_BYPASS _IOW('a',8,int)
+#define MVP_SET_AUD_CHANNEL _IOW('a',9,int)
+#define MVP_GET_AUD_STATUS _IOR('a',10,aud_status_t)
+#define MVP_SET_AUD_VOLUME _IOW('a',13,int)
+#define MVP_GET_AUD_VOLUME _IOR('a',14,int)
+#define MVP_SET_AUD_STREAMTYPE _IOW('a',15,int)
+#define MVP_SET_AUD_FORMAT _IOW('a',16,int)
+#define MVP_GET_AUD_SYNC _IOR('a',21,pts_sync_data_t*)
+#define MVP_SET_AUD_STC _IOW('a',22,long long int *)
+#define MVP_SET_AUD_SYNC _IOW('a',23,int)
+#define MVP_SET_AUD_END_STREAM _IOW('a',25,int)
+#define MVP_SET_AUD_RESET _IOW('a',26,int)
+#define MVP_SET_AUD_DAC_CLK _IOW('a',27,int)
+#define MVP_GET_AUD_REGS _IOW('a',28,aud_ctl_regs_t*)
+
+typedef struct _MvpData {
+ struct audio_output *audio_output;
+ struct audio_format audio_format;
+ int fd;
+} MvpData;
+
+static unsigned pcmfrequencies[][3] = {
+ {9, 8000, 32000},
+ {10, 11025, 44100},
+ {11, 12000, 48000},
+ {1, 16000, 32000},
+ {2, 22050, 44100},
+ {3, 24000, 48000},
+ {5, 32000, 32000},
+ {0, 44100, 44100},
+ {7, 48000, 48000},
+ {13, 64000, 32000},
+ {14, 88200, 44100},
+ {15, 96000, 48000}
+};
+
+static const unsigned numfrequencies =
+ sizeof(pcmfrequencies) / sizeof(pcmfrequencies[0]);
+
+static int mvp_testDefault(void)
+{
+ int fd;
+
+ fd = open("/dev/adec_pcm", O_WRONLY);
+
+ if (fd) {
+ close(fd);
+ return 0;
+ }
+
+ WARNING("Error opening PCM device \"/dev/adec_pcm\": %s\n",
+ strerror(errno));
+
+ return -1;
+}
+
+static void *mvp_initDriver(mpd_unused struct audio_output *audio_output,
+ mpd_unused const struct audio_format *audio_format,
+ mpd_unused ConfigParam *param)
+{
+ MvpData *md = xmalloc(sizeof(MvpData));
+ md->audio_output = audio_output;
+ md->fd = -1;
+
+ return md;
+}
+
+static void mvp_finishDriver(void *data)
+{
+ MvpData *md = data;
+ free(md);
+}
+
+static int mvp_setPcmParams(MvpData * md, unsigned long rate, int channels,
+ int big_endian, unsigned bits)
+{
+ unsigned iloop;
+ unsigned mix[5];
+
+ if (channels == 1)
+ mix[0] = 1;
+ else if (channels == 2)
+ mix[0] = 0;
+ else
+ return -1;
+
+ /* 0,1=24bit(24) , 2,3=16bit */
+ if (bits == 16)
+ mix[1] = 2;
+ else if (bits == 24)
+ mix[1] = 0;
+ else
+ return -1;
+
+ mix[3] = 0; /* stream type? */
+
+ if (big_endian == 1)
+ mix[4] = 1;
+ else if (big_endian == 0)
+ mix[4] = 0;
+ else
+ return -1;
+
+ /*
+ * if there is an exact match for the frequency, use it.
+ */
+ for (iloop = 0; iloop < numfrequencies; iloop++) {
+ if (rate == pcmfrequencies[iloop][1]) {
+ mix[2] = pcmfrequencies[iloop][0];
+ break;
+ }
+ }
+
+ if (iloop >= numfrequencies) {
+ ERROR("Can not find suitable output frequency for %ld\n", rate);
+ return -1;
+ }
+
+ if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
+ ERROR("Can not set audio format\n");
+ return -1;
+ }
+
+ if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) {
+ ERROR("Can not set audio sync\n");
+ return -1;
+ }
+
+ if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) {
+ ERROR("Can not set audio play mode\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+static int mvp_openDevice(void *data, struct audio_format *audioFormat)
+{
+ MvpData *md = data;
+ long long int stc = 0;
+ int mix[5] = { 0, 2, 7, 1, 0 };
+
+ if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) {
+ ERROR("Error opening /dev/adec_pcm: %s\n", strerror(errno));
+ return -1;
+ }
+ if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) {
+ ERROR("Error setting audio source: %s\n", strerror(errno));
+ return -1;
+ }
+ if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) {
+ ERROR("Error setting audio streamtype: %s\n", strerror(errno));
+ return -1;
+ }
+ if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
+ ERROR("Error setting audio format: %s\n", strerror(errno));
+ return -1;
+ }
+ ioctl(md->fd, MVP_SET_AUD_STC, &stc);
+ if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) {
+ ERROR("Error setting audio streamtype: %s\n", strerror(errno));
+ return -1;
+ }
+#ifdef WORDS_BIGENDIAN
+ mvp_setPcmParams(md, audioFormat->sample_rate, audioFormat->channels,
+ 0, audioFormat->bits);
+#else
+ mvp_setPcmParams(md, audioFormat->sample_rate, audioFormat->channels,
+ 1, audioFormat->bits);
+#endif
+ md->audio_format = *audioFormat;
+ return 0;
+}
+
+static void mvp_closeDevice(void *data)
+{
+ MvpData *md = data;
+ if (md->fd >= 0)
+ close(md->fd);
+ md->fd = -1;
+}
+
+static void mvp_dropBufferedAudio(void *data)
+{
+ MvpData *md = data;
+ if (md->fd >= 0) {
+ ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
+ close(md->fd);
+ md->fd = -1;
+ audio_output_closed(md->audio_output);
+ }
+}
+
+static int mvp_playAudio(void *data, const char *playChunk, size_t size)
+{
+ MvpData *md = data;
+ ssize_t ret;
+
+ /* reopen the device since it was closed by dropBufferedAudio */
+ if (md->fd < 0)
+ mvp_openDevice(md, &md->audio_format);
+
+ while (size > 0) {
+ ret = write(md->fd, playChunk, size);
+ if (ret < 0) {
+ if (errno == EINTR)
+ continue;
+ ERROR("closing mvp PCM device due to write error: "
+ "%s\n", strerror(errno));
+ mvp_closeDevice(md);
+ return -1;
+ }
+ playChunk += ret;
+ size -= ret;
+ }
+ return 0;
+}
+
+const struct audio_output_plugin mvpPlugin = {
+ .name = "mvp",
+ .test_default_device = mvp_testDefault,
+ .init = mvp_initDriver,
+ .finish = mvp_finishDriver,
+ .open = mvp_openDevice,
+ .play = mvp_playAudio,
+ .cancel = mvp_dropBufferedAudio,
+ .close = mvp_closeDevice,
+};
+
+#else /* HAVE_MVP */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(mvpPlugin)
+#endif /* HAVE_MVP */
diff --git a/src/output/null_plugin.c b/src/output/null_plugin.c
new file mode 100644
index 000000000..ff3a9833c
--- /dev/null
+++ b/src/output/null_plugin.c
@@ -0,0 +1,85 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+#include "../timer.h"
+#include "../utils.h"
+
+struct null_data {
+ Timer *timer;
+};
+
+static void *null_initDriver(mpd_unused struct audio_output *audioOutput,
+ mpd_unused const struct audio_format *audio_format,
+ mpd_unused ConfigParam *param)
+{
+ struct null_data *nd = xmalloc(sizeof(*nd));
+ nd->timer = NULL;
+ return nd;
+}
+
+static int null_openDevice(void *data,
+ struct audio_format *audio_format)
+{
+ struct null_data *nd = data;
+
+ nd->timer = timer_new(audio_format);
+ return 0;
+}
+
+static void null_closeDevice(void *data)
+{
+ struct null_data *nd = data;
+
+ if (nd->timer != NULL) {
+ timer_free(nd->timer);
+ nd->timer = NULL;
+ }
+}
+
+static int null_playAudio(void *data,
+ mpd_unused const char *playChunk, size_t size)
+{
+ struct null_data *nd = data;
+ Timer *timer = nd->timer;
+
+ if (!timer->started)
+ timer_start(timer);
+ else
+ timer_sync(timer);
+
+ timer_add(timer, size);
+
+ return 0;
+}
+
+static void null_dropBufferedAudio(void *data)
+{
+ struct null_data *nd = data;
+
+ timer_reset(nd->timer);
+}
+
+const struct audio_output_plugin nullPlugin = {
+ .name = "null",
+ .init = null_initDriver,
+ .open = null_openDevice,
+ .play = null_playAudio,
+ .cancel = null_dropBufferedAudio,
+ .close = null_closeDevice,
+};
diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c
new file mode 100644
index 000000000..67ea11fe4
--- /dev/null
+++ b/src/output/oss_plugin.c
@@ -0,0 +1,571 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * OSS audio output (c) 2004, 2005, 2006, 2007 by Eric Wong <eric@petta-tech.com>
+ * and Warren Dukes <warren.dukes@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+
+#ifdef HAVE_OSS
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+
+#if defined(__OpenBSD__) || defined(__NetBSD__)
+# include <soundcard.h>
+#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+# include <sys/soundcard.h>
+#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
+
+#ifdef WORDS_BIGENDIAN
+# define AFMT_S16_MPD AFMT_S16_BE
+#else
+# define AFMT_S16_MPD AFMT_S16_LE
+#endif /* WORDS_BIGENDIAN */
+
+typedef struct _OssData {
+ int fd;
+ const char *device;
+ struct audio_format audio_format;
+ int bitFormat;
+ int *supported[3];
+ int numSupported[3];
+ int *unsupported[3];
+ int numUnsupported[3];
+} OssData;
+
+enum oss_support {
+ OSS_SUPPORTED = 1,
+ OSS_UNSUPPORTED = 0,
+ OSS_UNKNOWN = -1,
+};
+
+enum oss_param {
+ OSS_RATE = 0,
+ OSS_CHANNELS = 1,
+ OSS_BITS = 2,
+};
+
+static enum oss_param
+getIndexForParam(unsigned param)
+{
+ enum oss_param idx = OSS_RATE;
+
+ switch (param) {
+ case SNDCTL_DSP_SPEED:
+ idx = OSS_RATE;
+ break;
+ case SNDCTL_DSP_CHANNELS:
+ idx = OSS_CHANNELS;
+ break;
+ case SNDCTL_DSP_SAMPLESIZE:
+ idx = OSS_BITS;
+ break;
+ }
+
+ return idx;
+}
+
+static int findSupportedParam(OssData * od, unsigned param, int val)
+{
+ int i;
+ enum oss_param idx = getIndexForParam(param);
+
+ for (i = 0; i < od->numSupported[idx]; i++) {
+ if (od->supported[idx][i] == val)
+ return 1;
+ }
+
+ return 0;
+}
+
+static int canConvert(int idx, int val)
+{
+ switch (idx) {
+ case OSS_BITS:
+ if (val != 16)
+ return 0;
+ break;
+ case OSS_CHANNELS:
+ if (val != 2)
+ return 0;
+ break;
+ }
+
+ return 1;
+}
+
+static int getSupportedParam(OssData * od, unsigned param, int val)
+{
+ int i;
+ enum oss_param idx = getIndexForParam(param);
+ int ret = -1;
+ int least = val;
+ int diff;
+
+ for (i = 0; i < od->numSupported[idx]; i++) {
+ diff = od->supported[idx][i] - val;
+ if (diff < 0)
+ diff = -diff;
+ if (diff < least) {
+ if (!canConvert(idx, od->supported[idx][i])) {
+ continue;
+ }
+ least = diff;
+ ret = od->supported[idx][i];
+ }
+ }
+
+ return ret;
+}
+
+static int findUnsupportedParam(OssData * od, unsigned param, int val)
+{
+ int i;
+ enum oss_param idx = getIndexForParam(param);
+
+ for (i = 0; i < od->numUnsupported[idx]; i++) {
+ if (od->unsupported[idx][i] == val)
+ return 1;
+ }
+
+ return 0;
+}
+
+static void addSupportedParam(OssData * od, unsigned param, int val)
+{
+ enum oss_param idx = getIndexForParam(param);
+
+ od->numSupported[idx]++;
+ od->supported[idx] = xrealloc(od->supported[idx],
+ od->numSupported[idx] * sizeof(int));
+ od->supported[idx][od->numSupported[idx] - 1] = val;
+}
+
+static void addUnsupportedParam(OssData * od, unsigned param, int val)
+{
+ enum oss_param idx = getIndexForParam(param);
+
+ od->numUnsupported[idx]++;
+ od->unsupported[idx] = xrealloc(od->unsupported[idx],
+ od->numUnsupported[idx] *
+ sizeof(int));
+ od->unsupported[idx][od->numUnsupported[idx] - 1] = val;
+}
+
+static void removeSupportedParam(OssData * od, unsigned param, int val)
+{
+ int i;
+ int j = 0;
+ enum oss_param idx = getIndexForParam(param);
+
+ for (i = 0; i < od->numSupported[idx] - 1; i++) {
+ if (od->supported[idx][i] == val)
+ j = 1;
+ od->supported[idx][i] = od->supported[idx][i + j];
+ }
+
+ od->numSupported[idx]--;
+ od->supported[idx] = xrealloc(od->supported[idx],
+ od->numSupported[idx] * sizeof(int));
+}
+
+static void removeUnsupportedParam(OssData * od, unsigned param, int val)
+{
+ int i;
+ int j = 0;
+ enum oss_param idx = getIndexForParam(param);
+
+ for (i = 0; i < od->numUnsupported[idx] - 1; i++) {
+ if (od->unsupported[idx][i] == val)
+ j = 1;
+ od->unsupported[idx][i] = od->unsupported[idx][i + j];
+ }
+
+ od->numUnsupported[idx]--;
+ od->unsupported[idx] = xrealloc(od->unsupported[idx],
+ od->numUnsupported[idx] *
+ sizeof(int));
+}
+
+static enum oss_support
+isSupportedParam(OssData * od, unsigned param, int val)
+{
+ if (findSupportedParam(od, param, val))
+ return OSS_SUPPORTED;
+ if (findUnsupportedParam(od, param, val))
+ return OSS_UNSUPPORTED;
+ return OSS_UNKNOWN;
+}
+
+static void supportParam(OssData * od, unsigned param, int val)
+{
+ enum oss_support supported = isSupportedParam(od, param, val);
+
+ if (supported == OSS_SUPPORTED)
+ return;
+
+ if (supported == OSS_UNSUPPORTED) {
+ removeUnsupportedParam(od, param, val);
+ }
+
+ addSupportedParam(od, param, val);
+}
+
+static void unsupportParam(OssData * od, unsigned param, int val)
+{
+ enum oss_support supported = isSupportedParam(od, param, val);
+
+ if (supported == OSS_UNSUPPORTED)
+ return;
+
+ if (supported == OSS_SUPPORTED) {
+ removeSupportedParam(od, param, val);
+ }
+
+ addUnsupportedParam(od, param, val);
+}
+
+static OssData *newOssData(void)
+{
+ OssData *ret = xmalloc(sizeof(OssData));
+
+ ret->device = NULL;
+ ret->fd = -1;
+
+ ret->supported[OSS_RATE] = NULL;
+ ret->supported[OSS_CHANNELS] = NULL;
+ ret->supported[OSS_BITS] = NULL;
+ ret->unsupported[OSS_RATE] = NULL;
+ ret->unsupported[OSS_CHANNELS] = NULL;
+ ret->unsupported[OSS_BITS] = NULL;
+
+ ret->numSupported[OSS_RATE] = 0;
+ ret->numSupported[OSS_CHANNELS] = 0;
+ ret->numSupported[OSS_BITS] = 0;
+ ret->numUnsupported[OSS_RATE] = 0;
+ ret->numUnsupported[OSS_CHANNELS] = 0;
+ ret->numUnsupported[OSS_BITS] = 0;
+
+ supportParam(ret, SNDCTL_DSP_SPEED, 48000);
+ supportParam(ret, SNDCTL_DSP_SPEED, 44100);
+ supportParam(ret, SNDCTL_DSP_CHANNELS, 2);
+ supportParam(ret, SNDCTL_DSP_SAMPLESIZE, 16);
+
+ return ret;
+}
+
+static void freeOssData(OssData * od)
+{
+ if (od->supported[OSS_RATE])
+ free(od->supported[OSS_RATE]);
+ if (od->supported[OSS_CHANNELS])
+ free(od->supported[OSS_CHANNELS]);
+ if (od->supported[OSS_BITS])
+ free(od->supported[OSS_BITS]);
+ if (od->unsupported[OSS_RATE])
+ free(od->unsupported[OSS_RATE]);
+ if (od->unsupported[OSS_CHANNELS])
+ free(od->unsupported[OSS_CHANNELS]);
+ if (od->unsupported[OSS_BITS])
+ free(od->unsupported[OSS_BITS]);
+
+ free(od);
+}
+
+#define OSS_STAT_NO_ERROR 0
+#define OSS_STAT_NOT_CHAR_DEV -1
+#define OSS_STAT_NO_PERMS -2
+#define OSS_STAT_DOESN_T_EXIST -3
+#define OSS_STAT_OTHER -4
+
+static int oss_statDevice(const char *device, int *stErrno)
+{
+ struct stat st;
+
+ if (0 == stat(device, &st)) {
+ if (!S_ISCHR(st.st_mode)) {
+ return OSS_STAT_NOT_CHAR_DEV;
+ }
+ } else {
+ *stErrno = errno;
+
+ switch (errno) {
+ case ENOENT:
+ case ENOTDIR:
+ return OSS_STAT_DOESN_T_EXIST;
+ case EACCES:
+ return OSS_STAT_NO_PERMS;
+ default:
+ return OSS_STAT_OTHER;
+ }
+ }
+
+ return 0;
+}
+
+static const char *default_devices[] = { "/dev/sound/dsp", "/dev/dsp" };
+
+static int oss_testDefault(void)
+{
+ int fd, i;
+
+ for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+ if ((fd = open(default_devices[i], O_WRONLY)) >= 0) {
+ xclose(fd);
+ return 0;
+ }
+ WARNING("Error opening OSS device \"%s\": %s\n",
+ default_devices[i], strerror(errno));
+ }
+
+ return -1;
+}
+
+static void *oss_open_default(ConfigParam *param)
+{
+ int i;
+ int err[ARRAY_SIZE(default_devices)];
+ int ret[ARRAY_SIZE(default_devices)];
+
+ for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+ ret[i] = oss_statDevice(default_devices[i], &err[i]);
+ if (ret[i] == 0) {
+ OssData *od = newOssData();
+ od->device = default_devices[i];
+ return od;
+ }
+ }
+
+ if (param)
+ ERROR("error trying to open specified OSS device"
+ " at line %i\n", param->line);
+ else
+ ERROR("error trying to open default OSS device\n");
+
+ for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
+ const char *dev = default_devices[i];
+ switch(ret[i]) {
+ case OSS_STAT_DOESN_T_EXIST:
+ ERROR("%s not found\n", dev);
+ break;
+ case OSS_STAT_NOT_CHAR_DEV:
+ ERROR("%s is not a character device\n", dev);
+ break;
+ case OSS_STAT_NO_PERMS:
+ ERROR("%s: permission denied\n", dev);
+ break;
+ default:
+ ERROR("Error accessing %s: %s\n", dev, strerror(err[i]));
+ }
+ }
+ exit(EXIT_FAILURE);
+ return NULL; /* some compilers can be dumb... */
+}
+
+static void *oss_initDriver(mpd_unused struct audio_output *audioOutput,
+ mpd_unused const struct audio_format *audio_format,
+ ConfigParam * param)
+{
+ if (param) {
+ BlockParam *bp = getBlockParam(param, "device");
+ if (bp) {
+ OssData *od = newOssData();
+ od->device = bp->value;
+ return od;
+ }
+ }
+ return oss_open_default(param);
+}
+
+static void oss_finishDriver(void *data)
+{
+ OssData *od = data;
+
+ freeOssData(od);
+}
+
+static int setParam(OssData * od, unsigned param, int *value)
+{
+ int val = *value;
+ int copy;
+ enum oss_support supported = isSupportedParam(od, param, val);
+
+ do {
+ if (supported == OSS_UNSUPPORTED) {
+ val = getSupportedParam(od, param, val);
+ if (copy < 0)
+ return -1;
+ }
+ copy = val;
+ if (ioctl(od->fd, param, &copy)) {
+ unsupportParam(od, param, val);
+ supported = OSS_UNSUPPORTED;
+ } else {
+ if (supported == OSS_UNKNOWN) {
+ supportParam(od, param, val);
+ supported = OSS_SUPPORTED;
+ }
+ val = copy;
+ }
+ } while (supported == OSS_UNSUPPORTED);
+
+ *value = val;
+
+ return 0;
+}
+
+static void oss_close(OssData * od)
+{
+ if (od->fd >= 0)
+ while (close(od->fd) && errno == EINTR) ;
+ od->fd = -1;
+}
+
+static int oss_open(OssData *od)
+{
+ int tmp;
+
+ if ((od->fd = open(od->device, O_WRONLY)) < 0) {
+ ERROR("Error opening OSS device \"%s\": %s\n", od->device,
+ strerror(errno));
+ goto fail;
+ }
+
+ tmp = od->audio_format.channels;
+ if (setParam(od, SNDCTL_DSP_CHANNELS, &tmp)) {
+ ERROR("OSS device \"%s\" does not support %u channels: %s\n",
+ od->device, od->audio_format.channels, strerror(errno));
+ goto fail;
+ }
+ od->audio_format.channels = tmp;
+
+ tmp = od->audio_format.sample_rate;
+ if (setParam(od, SNDCTL_DSP_SPEED, &tmp)) {
+ ERROR("OSS device \"%s\" does not support %u Hz audio: %s\n",
+ od->device, od->audio_format.sample_rate,
+ strerror(errno));
+ goto fail;
+ }
+ od->audio_format.sample_rate = tmp;
+
+ switch (od->audio_format.bits) {
+ case 8:
+ tmp = AFMT_S8;
+ break;
+ case 16:
+ tmp = AFMT_S16_MPD;
+ }
+
+ if (setParam(od, SNDCTL_DSP_SAMPLESIZE, &tmp)) {
+ ERROR("OSS device \"%s\" does not support %u bit audio: %s\n",
+ od->device, tmp, strerror(errno));
+ goto fail;
+ }
+
+ return 0;
+
+fail:
+ oss_close(od);
+ return -1;
+}
+
+static int oss_openDevice(void *data,
+ struct audio_format *audioFormat)
+{
+ int ret;
+ OssData *od = data;
+
+ od->audio_format = *audioFormat;
+
+ if ((ret = oss_open(od)) < 0)
+ return ret;
+
+ *audioFormat = od->audio_format;
+
+ DEBUG("oss device \"%s\" will be playing %u bit %u channel audio at "
+ "%u Hz\n", od->device,
+ od->audio_format.bits, od->audio_format.channels,
+ od->audio_format.sample_rate);
+
+ return ret;
+}
+
+static void oss_closeDevice(void *data)
+{
+ OssData *od = data;
+
+ oss_close(od);
+}
+
+static void oss_dropBufferedAudio(void *data)
+{
+ OssData *od = data;
+
+ if (od->fd >= 0) {
+ ioctl(od->fd, SNDCTL_DSP_RESET, 0);
+ oss_close(od);
+ }
+}
+
+static int oss_playAudio(void *data,
+ const char *playChunk, size_t size)
+{
+ OssData *od = data;
+ ssize_t ret;
+
+ /* reopen the device since it was closed by dropBufferedAudio */
+ if (od->fd < 0 && oss_open(od) < 0)
+ return -1;
+
+ while (size > 0) {
+ ret = write(od->fd, playChunk, size);
+ if (ret < 0) {
+ if (errno == EINTR)
+ continue;
+ ERROR("closing oss device \"%s\" due to write error: "
+ "%s\n", od->device, strerror(errno));
+ oss_closeDevice(od);
+ return -1;
+ }
+ playChunk += ret;
+ size -= ret;
+ }
+
+ return 0;
+}
+
+const struct audio_output_plugin ossPlugin = {
+ .name = "oss",
+ .test_default_device = oss_testDefault,
+ .init = oss_initDriver,
+ .finish = oss_finishDriver,
+ .open = oss_openDevice,
+ .play = oss_playAudio,
+ .cancel = oss_dropBufferedAudio,
+ .close = oss_closeDevice,
+};
+
+#else /* HAVE OSS */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(ossPlugin)
+#endif /* HAVE_OSS */
diff --git a/src/output/osx_plugin.c b/src/output/osx_plugin.c
new file mode 100644
index 000000000..65060cc8c
--- /dev/null
+++ b/src/output/osx_plugin.c
@@ -0,0 +1,368 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+
+#ifdef HAVE_OSX
+
+#include <AudioUnit/AudioUnit.h>
+
+#include "../utils.h"
+#include "../log.h"
+
+typedef struct _OsxData {
+ AudioUnit au;
+ pthread_mutex_t mutex;
+ pthread_cond_t condition;
+ char *buffer;
+ size_t bufferSize;
+ size_t pos;
+ size_t len;
+ int started;
+} OsxData;
+
+static OsxData *newOsxData()
+{
+ OsxData *ret = xmalloc(sizeof(OsxData));
+
+ pthread_mutex_init(&ret->mutex, NULL);
+ pthread_cond_init(&ret->condition, NULL);
+
+ ret->pos = 0;
+ ret->len = 0;
+ ret->started = 0;
+ ret->buffer = NULL;
+ ret->bufferSize = 0;
+
+ return ret;
+}
+
+static int osx_testDefault()
+{
+ /*AudioUnit au;
+ ComponentDescription desc;
+ Component comp;
+
+ desc.componentType = kAudioUnitType_Output;
+ desc.componentSubType = kAudioUnitSubType_Output;
+ desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+ desc.componentFlags = 0;
+ desc.componentFlagsMask = 0;
+
+ comp = FindNextComponent(NULL, &desc);
+ if(!comp) {
+ ERROR("Unable to open default OS X defice\n");
+ return -1;
+ }
+
+ if(OpenAComponent(comp, &au) != noErr) {
+ ERROR("Unable to open default OS X defice\n");
+ return -1;
+ }
+
+ CloseComponent(au); */
+
+ return 0;
+}
+
+static int osx_initDriver(struct audio_output *audioOutput,
+ mpd_unused const struct audio_format *audio_format,
+ ConfigParam * param)
+{
+ OsxData *od = newOsxData();
+
+ audioOutput->data = od;
+
+ return 0;
+}
+
+static void freeOsxData(OsxData * od)
+{
+ if (od->buffer)
+ free(od->buffer);
+ pthread_mutex_destroy(&od->mutex);
+ pthread_cond_destroy(&od->condition);
+ free(od);
+}
+
+static void osx_finishDriver(struct audio_output *audioOutput)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+ freeOsxData(od);
+}
+
+static void osx_dropBufferedAudio(struct audio_output *audioOutput)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+
+ pthread_mutex_lock(&od->mutex);
+ od->len = 0;
+ pthread_mutex_unlock(&od->mutex);
+}
+
+static void osx_closeDevice(struct audio_output *audioOutput)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+
+ pthread_mutex_lock(&od->mutex);
+ while (od->len) {
+ pthread_cond_wait(&od->condition, &od->mutex);
+ }
+ pthread_mutex_unlock(&od->mutex);
+
+ if (od->started) {
+ AudioOutputUnitStop(od->au);
+ od->started = 0;
+ }
+
+ CloseComponent(od->au);
+ AudioUnitUninitialize(od->au);
+}
+
+static OSStatus osx_render(void *vdata,
+ AudioUnitRenderActionFlags * ioActionFlags,
+ const AudioTimeStamp * inTimeStamp,
+ UInt32 inBusNumber, UInt32 inNumberFrames,
+ AudioBufferList * bufferList)
+{
+ OsxData *od = (OsxData *) vdata;
+ AudioBuffer *buffer = &bufferList->mBuffers[0];
+ size_t bufferSize = buffer->mDataByteSize;
+ size_t bytesToCopy;
+ int curpos = 0;
+
+ /*DEBUG("osx_render: enter : %i\n", (int)bufferList->mNumberBuffers);
+ DEBUG("osx_render: ioActionFlags: %p\n", ioActionFlags);
+ if(ioActionFlags) {
+ if(*ioActionFlags & kAudioUnitRenderAction_PreRender) {
+ DEBUG("prerender\n");
+ }
+ if(*ioActionFlags & kAudioUnitRenderAction_PostRender) {
+ DEBUG("post render\n");
+ }
+ if(*ioActionFlags & kAudioUnitRenderAction_OutputIsSilence) {
+ DEBUG("post render\n");
+ }
+ if(*ioActionFlags & kAudioOfflineUnitRenderAction_Preflight) {
+ DEBUG("prefilight\n");
+ }
+ if(*ioActionFlags & kAudioOfflineUnitRenderAction_Render) {
+ DEBUG("render\n");
+ }
+ if(*ioActionFlags & kAudioOfflineUnitRenderAction_Complete) {
+ DEBUG("complete\n");
+ }
+ } */
+
+ /* while(bufferSize) {
+ DEBUG("osx_render: lock\n"); */
+ pthread_mutex_lock(&od->mutex);
+ /*
+ DEBUG("%i:%i\n", bufferSize, od->len);
+ while(od->go && od->len < bufferSize &&
+ od->len < od->bufferSize)
+ {
+ DEBUG("osx_render: wait\n");
+ pthread_cond_wait(&od->condition, &od->mutex);
+ }
+ */
+
+ bytesToCopy = od->len < bufferSize ? od->len : bufferSize;
+ bufferSize = bytesToCopy;
+ od->len -= bytesToCopy;
+
+ if (od->pos + bytesToCopy > od->bufferSize) {
+ size_t bytes = od->bufferSize - od->pos;
+ memcpy(buffer->mData + curpos, od->buffer + od->pos, bytes);
+ od->pos = 0;
+ curpos += bytes;
+ bytesToCopy -= bytes;
+ }
+
+ memcpy(buffer->mData + curpos, od->buffer + od->pos, bytesToCopy);
+ od->pos += bytesToCopy;
+ curpos += bytesToCopy;
+
+ if (od->pos >= od->bufferSize)
+ od->pos = 0;
+ /* DEBUG("osx_render: unlock\n"); */
+ pthread_mutex_unlock(&od->mutex);
+ pthread_cond_signal(&od->condition);
+ /* } */
+
+ buffer->mDataByteSize = bufferSize;
+
+ if (!bufferSize) {
+ my_usleep(1000);
+ }
+
+ /* DEBUG("osx_render: leave\n"); */
+ return 0;
+}
+
+static int osx_openDevice(struct audio_output *audioOutput,
+ struct audio_format *audioFormat)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+ ComponentDescription desc;
+ Component comp;
+ AURenderCallbackStruct callback;
+ AudioStreamBasicDescription streamDesc;
+
+ desc.componentType = kAudioUnitType_Output;
+ desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+ desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+ desc.componentFlags = 0;
+ desc.componentFlagsMask = 0;
+
+ comp = FindNextComponent(NULL, &desc);
+ if (comp == 0) {
+ ERROR("Error finding OS X component\n");
+ return -1;
+ }
+
+ if (OpenAComponent(comp, &od->au) != noErr) {
+ ERROR("Unable to open OS X component\n");
+ return -1;
+ }
+
+ if (AudioUnitInitialize(od->au) != 0) {
+ CloseComponent(od->au);
+ ERROR("Unable to initialize OS X audio unit\n");
+ return -1;
+ }
+
+ callback.inputProc = osx_render;
+ callback.inputProcRefCon = od;
+
+ if (AudioUnitSetProperty(od->au, kAudioUnitProperty_SetRenderCallback,
+ kAudioUnitScope_Input, 0,
+ &callback, sizeof(callback)) != 0) {
+ AudioUnitUninitialize(od->au);
+ CloseComponent(od->au);
+ ERROR("unable to set callback for OS X audio unit\n");
+ return -1;
+ }
+
+ streamDesc.mSampleRate = audioFormat->sample_rate;
+ streamDesc.mFormatID = kAudioFormatLinearPCM;
+ streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+#ifdef WORDS_BIGENDIAN
+ streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+
+ streamDesc.mBytesPerPacket = audio_format_frame_size(audioFormat);
+ streamDesc.mFramesPerPacket = 1;
+ streamDesc.mBytesPerFrame = streamDesc.mBytesPerPacket;
+ streamDesc.mChannelsPerFrame = audioFormat->channels;
+ streamDesc.mBitsPerChannel = audioFormat->bits;
+
+ if (AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
+ kAudioUnitScope_Input, 0,
+ &streamDesc, sizeof(streamDesc)) != 0) {
+ AudioUnitUninitialize(od->au);
+ CloseComponent(od->au);
+ ERROR("Unable to set format on OS X device\n");
+ return -1;
+ }
+
+ /* create a buffer of 1s */
+ od->bufferSize = (audioFormat->sample_rate) *
+ audio_format_frame_size(audioFormat);
+ od->buffer = xrealloc(od->buffer, od->bufferSize);
+
+ od->pos = 0;
+ od->len = 0;
+
+ return 0;
+}
+
+static int osx_play(struct audio_output *audioOutput,
+ const char *playChunk, size_t size)
+{
+ OsxData *od = (OsxData *) audioOutput->data;
+ size_t bytesToCopy;
+ size_t curpos;
+
+ /* DEBUG("osx_play: enter\n"); */
+
+ if (!od->started) {
+ int err;
+ od->started = 1;
+ err = AudioOutputUnitStart(od->au);
+ if (err) {
+ ERROR("unable to start audio output: %i\n", err);
+ return -1;
+ }
+ }
+
+ pthread_mutex_lock(&od->mutex);
+
+ while (size) {
+ /* DEBUG("osx_play: lock\n"); */
+ curpos = od->pos + od->len;
+ if (curpos >= od->bufferSize)
+ curpos -= od->bufferSize;
+
+ bytesToCopy = od->bufferSize < size ? od->bufferSize : size;
+
+ while (od->len > od->bufferSize - bytesToCopy) {
+ /* DEBUG("osx_play: wait\n"); */
+ pthread_cond_wait(&od->condition, &od->mutex);
+ }
+
+ bytesToCopy = od->bufferSize - od->len;
+ bytesToCopy = bytesToCopy < size ? bytesToCopy : size;
+ size -= bytesToCopy;
+ od->len += bytesToCopy;
+
+ if (curpos + bytesToCopy > od->bufferSize) {
+ size_t bytes = od->bufferSize - curpos;
+ memcpy(od->buffer + curpos, playChunk, bytes);
+ curpos = 0;
+ playChunk += bytes;
+ bytesToCopy -= bytes;
+ }
+
+ memcpy(od->buffer + curpos, playChunk, bytesToCopy);
+ curpos += bytesToCopy;
+ playChunk += bytesToCopy;
+
+ }
+ /* DEBUG("osx_play: unlock\n"); */
+ pthread_mutex_unlock(&od->mutex);
+
+ /* DEBUG("osx_play: leave\n"); */
+ return 0;
+}
+
+const struct audio_output_plugin osxPlugin = {
+ .name = "osx",
+ .test_default_device = osx_testDefault,
+ .init = osx_initDriver,
+ .finish = osx_finishDriver,
+ .open = osx_openDevice,
+ .play = osx_play,
+ .cancel = osx_dropBufferedAudio,
+ .close = osx_closeDevice,
+};
+
+#else
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(osxPlugin)
+#endif
diff --git a/src/output/pulse_plugin.c b/src/output/pulse_plugin.c
new file mode 100644
index 000000000..b1ce3d049
--- /dev/null
+++ b/src/output/pulse_plugin.c
@@ -0,0 +1,218 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "../output_api.h"
+
+#ifdef HAVE_PULSE
+
+#include "../utils.h"
+#include "../log.h"
+
+#include <pulse/simple.h>
+#include <pulse/error.h>
+
+#define MPD_PULSE_NAME "mpd"
+#define CONN_ATTEMPT_INTERVAL 60
+
+typedef struct _PulseData {
+ struct audio_output *ao;
+
+ pa_simple *s;
+ char *server;
+ char *sink;
+ int connAttempts;
+ time_t lastAttempt;
+} PulseData;
+
+static PulseData *newPulseData(void)
+{
+ PulseData *ret;
+
+ ret = xmalloc(sizeof(PulseData));
+
+ ret->s = NULL;
+ ret->server = NULL;
+ ret->sink = NULL;
+ ret->connAttempts = 0;
+ ret->lastAttempt = 0;
+
+ return ret;
+}
+
+static void freePulseData(PulseData * pd)
+{
+ if (pd->server)
+ free(pd->server);
+ if (pd->sink)
+ free(pd->sink);
+ free(pd);
+}
+
+static void *pulse_initDriver(struct audio_output *ao,
+ mpd_unused const struct audio_format *audio_format,
+ ConfigParam * param)
+{
+ BlockParam *server = NULL;
+ BlockParam *sink = NULL;
+ PulseData *pd;
+
+ if (param) {
+ server = getBlockParam(param, "server");
+ sink = getBlockParam(param, "sink");
+ }
+
+ pd = newPulseData();
+ pd->ao = ao;
+ pd->server = server ? xstrdup(server->value) : NULL;
+ pd->sink = sink ? xstrdup(sink->value) : NULL;
+
+ return pd;
+}
+
+static void pulse_finishDriver(void *data)
+{
+ PulseData *pd = data;
+
+ freePulseData(pd);
+}
+
+static int pulse_testDefault(void)
+{
+ pa_simple *s;
+ pa_sample_spec ss;
+ int error;
+
+ ss.format = PA_SAMPLE_S16NE;
+ ss.rate = 44100;
+ ss.channels = 2;
+
+ s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL,
+ MPD_PULSE_NAME, &ss, NULL, NULL, &error);
+ if (!s) {
+ WARNING("Cannot connect to default PulseAudio server: %s\n",
+ pa_strerror(error));
+ return -1;
+ }
+
+ pa_simple_free(s);
+
+ return 0;
+}
+
+static int pulse_openDevice(void *data,
+ struct audio_format *audioFormat)
+{
+ PulseData *pd = data;
+ pa_sample_spec ss;
+ time_t t;
+ int error;
+
+ t = time(NULL);
+
+ if (pd->connAttempts != 0 &&
+ (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL)
+ return -1;
+
+ pd->connAttempts++;
+ pd->lastAttempt = t;
+
+ /* MPD doesn't support the other pulseaudio sample formats, so
+ we just force MPD to send us everything as 16 bit */
+ audioFormat->bits = 16;
+
+ ss.format = PA_SAMPLE_S16NE;
+ ss.rate = audioFormat->sample_rate;
+ ss.channels = audioFormat->channels;
+
+ pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
+ pd->sink, audio_output_get_name(pd->ao),
+ &ss, NULL, NULL,
+ &error);
+ if (!pd->s) {
+ ERROR("Cannot connect to server in PulseAudio output "
+ "\"%s\" (attempt %i): %s\n",
+ audio_output_get_name(pd->ao),
+ pd->connAttempts, pa_strerror(error));
+ return -1;
+ }
+
+ pd->connAttempts = 0;
+
+ DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i "
+ "channel audio at %i Hz\n",
+ audio_output_get_name(pd->ao),
+ audioFormat->bits,
+ audioFormat->channels, audioFormat->sample_rate);
+
+ return 0;
+}
+
+static void pulse_dropBufferedAudio(void *data)
+{
+ PulseData *pd = data;
+ int error;
+
+ if (pa_simple_flush(pd->s, &error) < 0)
+ WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
+ audio_output_get_name(pd->ao),
+ pa_strerror(error));
+}
+
+static void pulse_closeDevice(void *data)
+{
+ PulseData *pd = data;
+
+ if (pd->s) {
+ pa_simple_drain(pd->s, NULL);
+ pa_simple_free(pd->s);
+ }
+}
+
+static int pulse_playAudio(void *data,
+ const char *playChunk, size_t size)
+{
+ PulseData *pd = data;
+ int error;
+
+ if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
+ ERROR("PulseAudio output \"%s\" disconnecting due to write "
+ "error: %s\n",
+ audio_output_get_name(pd->ao),
+ pa_strerror(error));
+ pulse_closeDevice(pd);
+ return -1;
+ }
+
+ return 0;
+}
+
+const struct audio_output_plugin pulsePlugin = {
+ .name = "pulse",
+ .test_default_device = pulse_testDefault,
+ .init = pulse_initDriver,
+ .finish = pulse_finishDriver,
+ .open = pulse_openDevice,
+ .play = pulse_playAudio,
+ .cancel = pulse_dropBufferedAudio,
+ .close = pulse_closeDevice,
+};
+
+#else /* HAVE_PULSE */
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin)
+#endif /* HAVE_PULSE */
diff --git a/src/output/shout_mp3.c b/src/output/shout_mp3.c
new file mode 100644
index 000000000..e57b0c385
--- /dev/null
+++ b/src/output/shout_mp3.c
@@ -0,0 +1,188 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "shout_plugin.h"
+
+#ifdef HAVE_SHOUT_MP3
+
+#include "../utils.h"
+#include <lame/lame.h>
+
+struct lame_data {
+ lame_global_flags *gfp;
+};
+
+
+static int shout_mp3_encoder_init(struct shout_data *sd)
+{
+ struct lame_data *ld;
+
+ if (NULL == (ld = xmalloc(sizeof(*ld))))
+ FATAL("error initializing lame encoder data\n");
+ sd->encoder_data = ld;
+
+ return 0;
+}
+
+static int shout_mp3_encoder_clear_encoder(struct shout_data *sd)
+{
+ struct lame_data *ld = (struct lame_data *)sd->encoder_data;
+ struct shout_buffer *buf = &sd->buf;
+ int ret;
+
+ if ((ret = lame_encode_flush(ld->gfp, buf->data + buf->len,
+ buf->len)) < 0)
+ ERROR("error flushing lame buffers\n");
+
+ return (ret > 0);
+}
+
+static void shout_mp3_encoder_finish(struct shout_data *sd)
+{
+ struct lame_data *ld = (struct lame_data *)sd->encoder_data;
+
+ lame_close(ld->gfp);
+ ld->gfp = NULL;
+}
+
+static int shout_mp3_encoder_init_encoder(struct shout_data *sd)
+{
+ struct lame_data *ld = (struct lame_data *)sd->encoder_data;
+
+ if (NULL == (ld->gfp = lame_init())) {
+ ERROR("error initializing lame encoder for shout\n");
+ return -1;
+ }
+
+ if (sd->quality >= -1.0) {
+ if (0 != lame_set_VBR(ld->gfp, vbr_rh)) {
+ ERROR("error setting lame VBR mode\n");
+ return -1;
+ }
+ if (0 != lame_set_VBR_q(ld->gfp, sd->quality)) {
+ ERROR("error setting lame VBR quality\n");
+ return -1;
+ }
+ } else {
+ if (0 != lame_set_brate(ld->gfp, sd->bitrate)) {
+ ERROR("error setting lame bitrate\n");
+ return -1;
+ }
+ }
+
+ if (0 != lame_set_num_channels(ld->gfp,
+ sd->audio_format.channels)) {
+ ERROR("error setting lame num channels\n");
+ return -1;
+ }
+
+ if (0 != lame_set_in_samplerate(ld->gfp,
+ sd->audio_format.sample_rate)) {
+ ERROR("error setting lame sample rate\n");
+ return -1;
+ }
+
+ if (0 > lame_init_params(ld->gfp))
+ FATAL("error initializing lame params\n");
+
+ return 0;
+}
+
+static int shout_mp3_encoder_send_metadata(struct shout_data *sd,
+ char * song, size_t size)
+{
+ char artist[size];
+ char title[size];
+ int i;
+ struct tag *tag = sd->tag;
+
+ strncpy(artist, "", size);
+ strncpy(title, "", size);
+
+ for (i = 0; i < tag->numOfItems; i++) {
+ switch (tag->items[i]->type) {
+ case TAG_ITEM_ARTIST:
+ strncpy(artist, tag->items[i]->value, size);
+ break;
+ case TAG_ITEM_TITLE:
+ strncpy(title, tag->items[i]->value, size);
+ break;
+
+ default:
+ break;
+ }
+ }
+ snprintf(song, size, "%s - %s", title, artist);
+
+ return 1;
+}
+
+static int shout_mp3_encoder_encode(struct shout_data *sd,
+ const char * chunk, size_t len)
+{
+ unsigned int i;
+ int j;
+ float (*lamebuf)[2];
+ struct shout_buffer *buf = &(sd->buf);
+ unsigned int samples;
+ int bytes = audio_format_sample_size(&sd->audio_format);
+ struct lame_data *ld = (struct lame_data *)sd->encoder_data;
+ int bytes_out;
+
+ samples = len / (bytes * sd->audio_format.channels);
+ /* rough estimate, from lame.h */
+ lamebuf = xmalloc(sizeof(float) * (1.25 * samples + 7200));
+
+ /* this is for only 16-bit audio */
+
+ for (i = 0; i < samples; i++) {
+ for (j = 0; j < sd->audio_format.channels; j++) {
+ lamebuf[j][i] = *((const int16_t *) chunk);
+ chunk += bytes;
+ }
+ }
+
+ bytes_out = lame_encode_buffer_float(ld->gfp, lamebuf[0], lamebuf[1],
+ samples, buf->data,
+ sizeof(buf->data) - buf->len);
+ free(lamebuf);
+
+ if (0 > bytes_out) {
+ ERROR("error encoding lame buffer for shout\n");
+ lame_close(ld->gfp);
+ ld->gfp = NULL;
+ return -1;
+ } else
+ buf->len = bytes_out; /* signed to unsigned conversion */
+
+ return 0;
+}
+
+const struct shout_encoder_plugin shout_mp3_encoder = {
+ "mp3",
+ SHOUT_FORMAT_MP3,
+
+ shout_mp3_encoder_clear_encoder,
+ shout_mp3_encoder_encode,
+ shout_mp3_encoder_finish,
+ shout_mp3_encoder_init,
+ shout_mp3_encoder_init_encoder,
+ shout_mp3_encoder_send_metadata,
+};
+
+#endif
diff --git a/src/output/shout_ogg.c b/src/output/shout_ogg.c
new file mode 100644
index 000000000..248ffd1e8
--- /dev/null
+++ b/src/output/shout_ogg.c
@@ -0,0 +1,306 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "shout_plugin.h"
+
+#ifdef HAVE_SHOUT_OGG
+
+#include "../utils.h"
+#include <vorbis/vorbisenc.h>
+
+struct ogg_vorbis_data {
+ ogg_stream_state os;
+ ogg_page og;
+ ogg_packet op;
+ ogg_packet header_main;
+ ogg_packet header_comments;
+ ogg_packet header_codebooks;
+
+ vorbis_dsp_state vd;
+ vorbis_block vb;
+ vorbis_info vi;
+ vorbis_comment vc;
+};
+
+static void add_tag(struct ogg_vorbis_data *od, const char *name, char *value)
+{
+ if (value) {
+ union {
+ const char *in;
+ char *out;
+ } u = { .in = name };
+ vorbis_comment_add_tag(&od->vc, u.out, value);
+ }
+}
+
+static void copy_tag_to_vorbis_comment(struct shout_data *sd)
+{
+ struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+ if (sd->tag) {
+ int i;
+
+ for (i = 0; i < sd->tag->numOfItems; i++) {
+ switch (sd->tag->items[i]->type) {
+ case TAG_ITEM_ARTIST:
+ add_tag(od, "ARTIST", sd->tag->items[i]->value);
+ break;
+ case TAG_ITEM_ALBUM:
+ add_tag(od, "ALBUM", sd->tag->items[i]->value);
+ break;
+ case TAG_ITEM_TITLE:
+ add_tag(od, "TITLE", sd->tag->items[i]->value);
+ break;
+
+ default:
+ break;
+ }
+ }
+ }
+}
+
+static int copy_ogg_buffer_to_shout_buffer(ogg_page *og,
+ struct shout_buffer *buf)
+{
+ if (sizeof(buf->data) - buf->len >= (size_t)og->header_len) {
+ memcpy(buf->data + buf->len,
+ og->header, og->header_len);
+ buf->len += og->header_len;
+ } else {
+ ERROR("%s: not enough buffer space!\n", __func__);
+ return -1;
+ }
+
+ if (sizeof(buf->data) - buf->len >= (size_t)og->body_len) {
+ memcpy(buf->data + buf->len,
+ og->body, og->body_len);
+ buf->len += og->body_len;
+ } else {
+ ERROR("%s: not enough buffer space!\n", __func__);
+ return -1;
+ }
+
+ return 0;
+}
+
+static int flush_ogg_buffer(struct shout_data *sd)
+{
+ struct shout_buffer *buf = &sd->buf;
+ struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+ int ret = 0;
+
+ if (ogg_stream_flush(&od->os, &od->og))
+ ret = copy_ogg_buffer_to_shout_buffer(&od->og, buf);
+
+ return ret;
+}
+
+static int send_ogg_vorbis_header(struct shout_data *sd)
+{
+ struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+ vorbis_analysis_headerout(&od->vd, &od->vc,
+ &od->header_main,
+ &od->header_comments,
+ &od->header_codebooks);
+
+ ogg_stream_packetin(&od->os, &od->header_main);
+ ogg_stream_packetin(&od->os, &od->header_comments);
+ ogg_stream_packetin(&od->os, &od->header_codebooks);
+
+ return flush_ogg_buffer(sd);
+}
+
+static void finish_encoder(struct ogg_vorbis_data *od)
+{
+ vorbis_analysis_wrote(&od->vd, 0);
+
+ while (vorbis_analysis_blockout(&od->vd, &od->vb) == 1) {
+ vorbis_analysis(&od->vb, NULL);
+ vorbis_bitrate_addblock(&od->vb);
+ while (vorbis_bitrate_flushpacket(&od->vd, &od->op)) {
+ ogg_stream_packetin(&od->os, &od->op);
+ }
+ }
+}
+
+static int shout_ogg_encoder_clear_encoder(struct shout_data *sd)
+{
+ struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+ int ret;
+
+ finish_encoder(od);
+ if ((ret = ogg_stream_pageout(&od->os, &od->og)))
+ copy_ogg_buffer_to_shout_buffer(&od->og, &sd->buf);
+
+ vorbis_comment_clear(&od->vc);
+ ogg_stream_clear(&od->os);
+ vorbis_block_clear(&od->vb);
+ vorbis_dsp_clear(&od->vd);
+ vorbis_info_clear(&od->vi);
+
+ return ret;
+}
+
+static void shout_ogg_encoder_finish(struct shout_data *sd)
+{
+ struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+ if (od) {
+ free(od);
+ sd->encoder_data = NULL;
+ }
+}
+
+static int shout_ogg_encoder_init(struct shout_data *sd)
+{
+ struct ogg_vorbis_data *od;
+
+ if (NULL == (od = xmalloc(sizeof(*od))))
+ FATAL("error initializing ogg vorbis encoder data\n");
+ sd->encoder_data = od;
+
+ return 0;
+}
+
+static int reinit_encoder(struct shout_data *sd)
+{
+ struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+ vorbis_info_init(&od->vi);
+
+ if (sd->quality >= -1.0) {
+ if (0 != vorbis_encode_init_vbr(&od->vi,
+ sd->audio_format.channels,
+ sd->audio_format.sample_rate,
+ sd->quality * 0.1)) {
+ ERROR("error initializing vorbis vbr\n");
+ vorbis_info_clear(&od->vi);
+ return -1;
+ }
+ } else {
+ if (0 != vorbis_encode_init(&od->vi,
+ sd->audio_format.channels,
+ sd->audio_format.sample_rate, -1.0,
+ sd->bitrate * 1000, -1.0)) {
+ ERROR("error initializing vorbis encoder\n");
+ vorbis_info_clear(&od->vi);
+ return -1;
+ }
+ }
+
+ vorbis_analysis_init(&od->vd, &od->vi);
+ vorbis_block_init(&od->vd, &od->vb);
+ ogg_stream_init(&od->os, rand());
+ vorbis_comment_init(&od->vc);
+
+ return 0;
+}
+
+static int shout_ogg_encoder_init_encoder(struct shout_data *sd)
+{
+ if (reinit_encoder(sd))
+ return -1;
+
+ if (send_ogg_vorbis_header(sd)) {
+ ERROR("error sending ogg vorbis header for shout\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+static int shout_ogg_encoder_send_metadata(struct shout_data *sd,
+ mpd_unused char * song,
+ mpd_unused size_t size)
+{
+ struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+ shout_ogg_encoder_clear_encoder(sd);
+ if (reinit_encoder(sd))
+ return 0;
+
+ copy_tag_to_vorbis_comment(sd);
+
+ vorbis_analysis_headerout(&od->vd, &od->vc,
+ &od->header_main,
+ &od->header_comments,
+ &od->header_codebooks);
+
+ ogg_stream_packetin(&od->os, &od->header_main);
+ ogg_stream_packetin(&od->os, &od->header_comments);
+ ogg_stream_packetin(&od->os, &od->header_codebooks);
+
+ flush_ogg_buffer(sd);
+
+ return 0;
+}
+
+static int shout_ogg_encoder_encode(struct shout_data *sd,
+ const char *chunk, size_t size)
+{
+ struct shout_buffer *buf = &sd->buf;
+ unsigned int i;
+ int j;
+ float **vorbbuf;
+ unsigned int samples;
+ int bytes = audio_format_sample_size(&sd->audio_format);
+ struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
+
+ samples = size / (bytes * sd->audio_format.channels);
+ vorbbuf = vorbis_analysis_buffer(&od->vd, samples);
+
+ /* this is for only 16-bit audio */
+
+ for (i = 0; i < samples; i++) {
+ for (j = 0; j < sd->audio_format.channels; j++) {
+ vorbbuf[j][i] = (*((const int16_t *) chunk)) / 32768.0;
+ chunk += bytes;
+ }
+ }
+
+ vorbis_analysis_wrote(&od->vd, samples);
+
+ while (1 == vorbis_analysis_blockout(&od->vd, &od->vb)) {
+ vorbis_analysis(&od->vb, NULL);
+ vorbis_bitrate_addblock(&od->vb);
+
+ while (vorbis_bitrate_flushpacket(&od->vd, &od->op)) {
+ ogg_stream_packetin(&od->os, &od->op);
+ }
+ }
+
+ if (ogg_stream_pageout(&od->os, &od->og))
+ copy_ogg_buffer_to_shout_buffer(&od->og, buf);
+
+ return 0;
+}
+
+const struct shout_encoder_plugin shout_ogg_encoder = {
+ "ogg",
+ SHOUT_FORMAT_VORBIS,
+
+ shout_ogg_encoder_clear_encoder,
+ shout_ogg_encoder_encode,
+ shout_ogg_encoder_finish,
+ shout_ogg_encoder_init,
+ shout_ogg_encoder_init_encoder,
+ shout_ogg_encoder_send_metadata,
+};
+
+#endif
diff --git a/src/output/shout_plugin.c b/src/output/shout_plugin.c
new file mode 100644
index 000000000..dd909bdae
--- /dev/null
+++ b/src/output/shout_plugin.c
@@ -0,0 +1,596 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "shout_plugin.h"
+
+#ifdef HAVE_SHOUT
+
+#include "../utils.h"
+
+#include <assert.h>
+
+#define CONN_ATTEMPT_INTERVAL 60
+#define DEFAULT_CONN_TIMEOUT 2
+
+static int shout_init_count;
+
+static const struct shout_encoder_plugin *const shout_encoder_plugins[] = {
+#ifdef HAVE_SHOUT_MP3
+ &shout_mp3_encoder,
+#endif
+#ifdef HAVE_SHOUT_OGG
+ &shout_ogg_encoder,
+#endif
+ NULL
+};
+
+static const struct shout_encoder_plugin *
+shout_encoder_plugin_get(const char *name)
+{
+ unsigned i;
+
+ for (i = 0; shout_encoder_plugins[i] != NULL; ++i)
+ if (strcmp(shout_encoder_plugins[i]->name, name) == 0)
+ return shout_encoder_plugins[i];
+
+ return NULL;
+}
+
+static struct shout_data *new_shout_data(void)
+{
+ struct shout_data *ret = xmalloc(sizeof(*ret));
+
+ ret->shout_conn = shout_new();
+ ret->shout_meta = shout_metadata_new();
+ ret->opened = 0;
+ ret->tag = NULL;
+ ret->tag_to_send = 0;
+ ret->bitrate = -1;
+ ret->quality = -2.0;
+ ret->timeout = DEFAULT_CONN_TIMEOUT;
+ ret->conn_attempts = 0;
+ ret->last_attempt = 0;
+ ret->timer = NULL;
+ ret->buf.len = 0;
+
+ return ret;
+}
+
+static void free_shout_data(struct shout_data *sd)
+{
+ if (sd->shout_meta)
+ shout_metadata_free(sd->shout_meta);
+ if (sd->shout_conn)
+ shout_free(sd->shout_conn);
+ if (sd->tag)
+ tag_free(sd->tag);
+ if (sd->timer)
+ timer_free(sd->timer);
+
+ free(sd);
+}
+
+#define check_block_param(name) { \
+ block_param = getBlockParam(param, name); \
+ if (!block_param) { \
+ FATAL("no \"%s\" defined for shout device defined at line " \
+ "%i\n", name, param->line); \
+ } \
+ }
+
+static void *my_shout_init_driver(struct audio_output *audio_output,
+ const struct audio_format *audio_format,
+ ConfigParam *param)
+{
+ struct shout_data *sd;
+ char *test;
+ int port;
+ char *host;
+ char *mount;
+ char *passwd;
+ const char *encoding;
+ unsigned protocol;
+ const char *user;
+ char *name;
+ BlockParam *block_param;
+ int public;
+
+ sd = new_shout_data();
+ sd->audio_output = audio_output;
+
+ if (shout_init_count == 0)
+ shout_init();
+
+ shout_init_count++;
+
+ check_block_param("host");
+ host = block_param->value;
+
+ check_block_param("mount");
+ mount = block_param->value;
+
+ check_block_param("port");
+
+ port = strtol(block_param->value, &test, 10);
+
+ if (*test != '\0' || port <= 0) {
+ FATAL("shout port \"%s\" is not a positive integer, line %i\n",
+ block_param->value, block_param->line);
+ }
+
+ check_block_param("password");
+ passwd = block_param->value;
+
+ check_block_param("name");
+ name = block_param->value;
+
+ public = getBoolBlockParam(param, "public", 1);
+ if (public == CONF_BOOL_UNSET)
+ public = 0;
+
+ block_param = getBlockParam(param, "user");
+ if (block_param)
+ user = block_param->value;
+ else
+ user = "source";
+
+ block_param = getBlockParam(param, "quality");
+
+ if (block_param) {
+ int line = block_param->line;
+
+ sd->quality = strtod(block_param->value, &test);
+
+ if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
+ FATAL("shout quality \"%s\" is not a number in the "
+ "range -1 to 10, line %i\n", block_param->value,
+ block_param->line);
+ }
+
+ block_param = getBlockParam(param, "bitrate");
+
+ if (block_param) {
+ FATAL("quality (line %i) and bitrate (line %i) are "
+ "both defined for shout output\n", line,
+ block_param->line);
+ }
+ } else {
+ block_param = getBlockParam(param, "bitrate");
+
+ if (!block_param) {
+ FATAL("neither bitrate nor quality defined for shout "
+ "output at line %i\n", param->line);
+ }
+
+ sd->bitrate = strtol(block_param->value, &test, 10);
+
+ if (*test != '\0' || sd->bitrate <= 0) {
+ FATAL("bitrate at line %i should be a positive integer "
+ "\n", block_param->line);
+ }
+ }
+
+ check_block_param("format");
+
+ assert(audio_format != NULL);
+ sd->audio_format = *audio_format;
+
+ block_param = getBlockParam(param, "encoding");
+ if (block_param) {
+ if (0 == strcmp(block_param->value, "mp3"))
+ encoding = block_param->value;
+ else if (0 == strcmp(block_param->value, "ogg"))
+ encoding = block_param->value;
+ else
+ FATAL("shout encoding \"%s\" is not \"ogg\" or "
+ "\"mp3\", line %i\n", block_param->value,
+ block_param->line);
+ } else {
+ encoding = "ogg";
+ }
+
+ sd->encoder = shout_encoder_plugin_get(encoding);
+ if (sd->encoder == NULL)
+ FATAL("couldn't find shout encoder plugin for \"%s\" "
+ "at line %i\n", encoding, block_param->line);
+
+ check_block_param("protocol");
+
+ block_param = getBlockParam(param, "protocol");
+ if (block_param) {
+ if (0 == strcmp(block_param->value, "shoutcast") &&
+ 0 != strcmp(encoding, "mp3"))
+ FATAL("you cannot stream \"%s\" to shoutcast, use mp3\n",
+ encoding);
+ else if (0 == strcmp(block_param->value, "shoutcast"))
+ protocol = SHOUT_PROTOCOL_ICY;
+ else if (0 == strcmp(block_param->value, "icecast1"))
+ protocol = SHOUT_PROTOCOL_XAUDIOCAST;
+ else if (0 == strcmp(block_param->value, "icecast2"))
+ protocol = SHOUT_PROTOCOL_HTTP;
+ else
+ FATAL("shout protocol \"%s\" is not \"shoutcast\" or "
+ "\"icecast1\"or "
+ "\"icecast2\", line %i\n", block_param->value,
+ block_param->line);
+ } else {
+ protocol = SHOUT_PROTOCOL_HTTP;
+ }
+
+ if (shout_set_host(sd->shout_conn, host) != SHOUTERR_SUCCESS ||
+ shout_set_port(sd->shout_conn, port) != SHOUTERR_SUCCESS ||
+ shout_set_password(sd->shout_conn, passwd) != SHOUTERR_SUCCESS ||
+ shout_set_mount(sd->shout_conn, mount) != SHOUTERR_SUCCESS ||
+ shout_set_name(sd->shout_conn, name) != SHOUTERR_SUCCESS ||
+ shout_set_user(sd->shout_conn, user) != SHOUTERR_SUCCESS ||
+ shout_set_public(sd->shout_conn, public) != SHOUTERR_SUCCESS ||
+ shout_set_nonblocking(sd->shout_conn, 1) != SHOUTERR_SUCCESS ||
+ shout_set_format(sd->shout_conn, sd->encoder->shout_format)
+ != SHOUTERR_SUCCESS ||
+ shout_set_protocol(sd->shout_conn, protocol) != SHOUTERR_SUCCESS ||
+ shout_set_agent(sd->shout_conn, "MPD") != SHOUTERR_SUCCESS) {
+ FATAL("error configuring shout defined at line %i: %s\n",
+ param->line, shout_get_error(sd->shout_conn));
+ }
+
+ /* optional paramters */
+ block_param = getBlockParam(param, "timeout");
+ if (block_param) {
+ sd->timeout = (int)strtol(block_param->value, &test, 10);
+ if (*test != '\0' || sd->timeout <= 0) {
+ FATAL("shout timeout is not a positive integer, "
+ "line %i\n", block_param->line);
+ }
+ }
+
+ block_param = getBlockParam(param, "genre");
+ if (block_param && shout_set_genre(sd->shout_conn, block_param->value)) {
+ FATAL("error configuring shout defined at line %i: %s\n",
+ param->line, shout_get_error(sd->shout_conn));
+ }
+
+ block_param = getBlockParam(param, "description");
+ if (block_param && shout_set_description(sd->shout_conn,
+ block_param->value)) {
+ FATAL("error configuring shout defined at line %i: %s\n",
+ param->line, shout_get_error(sd->shout_conn));
+ }
+
+ {
+ char temp[11];
+ memset(temp, 0, sizeof(temp));
+
+ snprintf(temp, sizeof(temp), "%u", sd->audio_format.channels);
+ shout_set_audio_info(sd->shout_conn, SHOUT_AI_CHANNELS, temp);
+
+ snprintf(temp, sizeof(temp), "%u", sd->audio_format.sample_rate);
+
+ shout_set_audio_info(sd->shout_conn, SHOUT_AI_SAMPLERATE, temp);
+
+ if (sd->quality >= -1.0) {
+ snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
+ shout_set_audio_info(sd->shout_conn, SHOUT_AI_QUALITY,
+ temp);
+ } else {
+ snprintf(temp, sizeof(temp), "%d", sd->bitrate);
+ shout_set_audio_info(sd->shout_conn, SHOUT_AI_BITRATE,
+ temp);
+ }
+ }
+
+ if (sd->encoder->init_func(sd) != 0)
+ FATAL("shout: encoder plugin '%s' failed to initialize\n",
+ sd->encoder->name);
+
+ return sd;
+}
+
+static int handle_shout_error(struct shout_data *sd, int err)
+{
+ switch (err) {
+ case SHOUTERR_SUCCESS:
+ break;
+ case SHOUTERR_UNCONNECTED:
+ case SHOUTERR_SOCKET:
+ ERROR("Lost shout connection to %s:%i: %s\n",
+ shout_get_host(sd->shout_conn),
+ shout_get_port(sd->shout_conn),
+ shout_get_error(sd->shout_conn));
+ sd->shout_error = 1;
+ return -1;
+ default:
+ ERROR("shout: connection to %s:%i error: %s\n",
+ shout_get_host(sd->shout_conn),
+ shout_get_port(sd->shout_conn),
+ shout_get_error(sd->shout_conn));
+ sd->shout_error = 1;
+ return -1;
+ }
+
+ return 0;
+}
+
+static int write_page(struct shout_data *sd)
+{
+ int err;
+
+ if (sd->buf.len == 0)
+ return 0;
+
+ shout_sync(sd->shout_conn);
+ err = shout_send(sd->shout_conn, sd->buf.data, sd->buf.len);
+ if (handle_shout_error(sd, err) < 0)
+ return -1;
+ sd->buf.len = 0;
+
+ return 0;
+}
+
+static void close_shout_conn(struct shout_data * sd)
+{
+ if (sd->opened) {
+ if (sd->encoder->clear_encoder_func(sd))
+ write_page(sd);
+ }
+
+ if (shout_get_connected(sd->shout_conn) != SHOUTERR_UNCONNECTED &&
+ shout_close(sd->shout_conn) != SHOUTERR_SUCCESS) {
+ ERROR("problem closing connection to shout server: %s\n",
+ shout_get_error(sd->shout_conn));
+ }
+
+ sd->opened = 0;
+}
+
+static void my_shout_finish_driver(void *data)
+{
+ struct shout_data *sd = (struct shout_data *)data;
+
+ close_shout_conn(sd);
+
+ sd->encoder->finish_func(sd);
+ free_shout_data(sd);
+
+ shout_init_count--;
+
+ if (shout_init_count == 0)
+ shout_shutdown();
+}
+
+static void my_shout_drop_buffered_audio(void *data)
+{
+ struct shout_data *sd = (struct shout_data *)data;
+ timer_reset(sd->timer);
+
+ /* needs to be implemented for shout */
+}
+
+static void my_shout_close_device(void *data)
+{
+ struct shout_data *sd = (struct shout_data *)data;
+
+ close_shout_conn(sd);
+
+ if (sd->timer) {
+ timer_free(sd->timer);
+ sd->timer = NULL;
+ }
+}
+
+static int shout_connect(struct shout_data *sd)
+{
+ time_t t = time(NULL);
+ int state = shout_get_connected(sd->shout_conn);
+
+ /* already connected */
+ if (state == SHOUTERR_CONNECTED)
+ return 0;
+
+ /* waiting to connect */
+ if (state == SHOUTERR_BUSY && sd->conn_attempts != 0) {
+ /* timeout waiting to connect */
+ if ((t - sd->last_attempt) > sd->timeout) {
+ ERROR("timeout connecting to shout server %s:%i "
+ "(attempt %i)\n",
+ shout_get_host(sd->shout_conn),
+ shout_get_port(sd->shout_conn),
+ sd->conn_attempts);
+ return -1;
+ }
+
+ return 1;
+ }
+
+ /* we're in some funky state, so just reset it to unconnected */
+ if (state != SHOUTERR_UNCONNECTED)
+ shout_close(sd->shout_conn);
+
+ /* throttle new connection attempts */
+ if (sd->conn_attempts != 0 &&
+ (t - sd->last_attempt) <= CONN_ATTEMPT_INTERVAL) {
+ return -1;
+ }
+
+ /* initiate a new connection */
+
+ sd->conn_attempts++;
+ sd->last_attempt = t;
+
+ state = shout_open(sd->shout_conn);
+ switch (state) {
+ case SHOUTERR_SUCCESS:
+ case SHOUTERR_CONNECTED:
+ return 0;
+ case SHOUTERR_BUSY:
+ return 1;
+ default:
+ ERROR("problem opening connection to shout server %s:%i "
+ "(attempt %i): %s\n",
+ shout_get_host(sd->shout_conn),
+ shout_get_port(sd->shout_conn),
+ sd->conn_attempts, shout_get_error(sd->shout_conn));
+ return -1;
+ }
+}
+
+static int open_shout_conn(void *data)
+{
+ struct shout_data *sd = (struct shout_data *)data;
+ int status;
+
+ status = shout_connect(sd);
+ if (status != 0)
+ return status;
+
+ if (sd->encoder->init_encoder_func(sd) < 0) {
+ shout_close(sd->shout_conn);
+ return -1;
+ }
+
+ write_page(sd);
+
+ sd->shout_error = 0;
+ sd->opened = 1;
+ sd->tag_to_send = 1;
+ sd->conn_attempts = 0;
+
+ return 0;
+}
+
+static int my_shout_open_device(void *data,
+ struct audio_format *audio_format)
+{
+ struct shout_data *sd = (struct shout_data *)data;
+
+ if (!sd->opened && open_shout_conn(sd) < 0)
+ return -1;
+
+ if (sd->timer)
+ timer_free(sd->timer);
+
+ sd->timer = timer_new(audio_format);
+
+ return 0;
+}
+
+static void send_metadata(struct shout_data * sd)
+{
+ static const int size = 1024;
+ char song[size];
+
+ if (!sd->opened || !sd->tag)
+ return;
+
+ if (sd->encoder->send_metadata_func(sd, song, size)) {
+ shout_metadata_add(sd->shout_meta, "song", song);
+ if (SHOUTERR_SUCCESS != shout_set_metadata(sd->shout_conn,
+ sd->shout_meta)) {
+ ERROR("error setting shout metadata\n");
+ return;
+ }
+ }
+
+ sd->tag_to_send = 0;
+}
+
+static int my_shout_play(void *data,
+ const char *chunk, size_t size)
+{
+ struct shout_data *sd = (struct shout_data *)data;
+ int status;
+
+ if (!sd->timer->started)
+ timer_start(sd->timer);
+
+ timer_add(sd->timer, size);
+
+ if (sd->opened && sd->tag_to_send)
+ send_metadata(sd);
+
+ if (!sd->opened) {
+ status = open_shout_conn(sd);
+ if (status < 0) {
+ my_shout_close_device(sd);
+ return -1;
+ } else if (status > 0) {
+ timer_sync(sd->timer);
+ return 0;
+ }
+ }
+
+ if (sd->encoder->encode_func(sd, chunk, size)) {
+ my_shout_close_device(sd);
+ return -1;
+ }
+
+ if (write_page(sd) < 0) {
+ my_shout_close_device(sd);
+ return -1;
+ }
+
+ return 0;
+}
+
+static void my_shout_pause(void *data)
+{
+ struct shout_data *sd = (struct shout_data *)data;
+ static const char silence[1020];
+ int ret;
+
+ /* play silence until the player thread sends us a command */
+
+ while (sd->opened && !audio_output_is_pending(sd->audio_output)) {
+ ret = my_shout_play(data, silence, sizeof(silence));
+ if (ret != 0)
+ break;
+ }
+}
+
+static void my_shout_set_tag(void *data,
+ const struct tag *tag)
+{
+ struct shout_data *sd = (struct shout_data *)data;
+
+ if (sd->tag)
+ tag_free(sd->tag);
+ sd->tag = NULL;
+ sd->tag_to_send = 0;
+
+ if (!tag)
+ return;
+
+ sd->tag = tag_dup(tag);
+ sd->tag_to_send = 1;
+}
+
+const struct audio_output_plugin shoutPlugin = {
+ .name = "shout",
+ .init = my_shout_init_driver,
+ .finish = my_shout_finish_driver,
+ .open = my_shout_open_device,
+ .play = my_shout_play,
+ .pause = my_shout_pause,
+ .cancel = my_shout_drop_buffered_audio,
+ .close = my_shout_close_device,
+ .send_tag = my_shout_set_tag,
+};
+
+#else
+
+DISABLED_AUDIO_OUTPUT_PLUGIN(shoutPlugin)
+#endif
diff --git a/src/output/shout_plugin.h b/src/output/shout_plugin.h
new file mode 100644
index 000000000..2cfe68f29
--- /dev/null
+++ b/src/output/shout_plugin.h
@@ -0,0 +1,93 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * This project's homepage is: http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef AUDIO_OUTPUT_SHOUT_H
+#define AUDIO_OUTPUT_SHOUT_H
+
+#include "../output_api.h"
+
+#ifdef HAVE_SHOUT
+
+#include "../conf.h"
+#include "../timer.h"
+
+#include <shout/shout.h>
+
+struct shout_data;
+
+struct shout_encoder_plugin {
+ const char *name;
+ unsigned int shout_format;
+
+ int (*clear_encoder_func)(struct shout_data *sd);
+ int (*encode_func)(struct shout_data *sd,
+ const char *chunk, size_t len);
+ void (*finish_func)(struct shout_data *sd);
+ int (*init_func)(struct shout_data *sd);
+ int (*init_encoder_func) (struct shout_data *sd);
+ /* Called when there is a new MpdTag to encode into the
+ stream. If this function returns non-zero, then the
+ resulting song will be passed to the shout server as
+ metadata. This allows the Ogg encoder to send metadata via
+ Vorbis comments in the stream, while an MP3 encoder can use
+ the Shout Server's metadata API. */
+ int (*send_metadata_func)(struct shout_data *sd,
+ char *song, size_t size);
+};
+
+struct shout_buffer {
+ unsigned char data[8192];
+ size_t len;
+};
+
+struct shout_data {
+ struct audio_output *audio_output;
+
+ shout_t *shout_conn;
+ shout_metadata_t *shout_meta;
+ int shout_error;
+
+ const struct shout_encoder_plugin *encoder;
+ void *encoder_data;
+
+ float quality;
+ int bitrate;
+
+ int opened;
+
+ struct tag *tag;
+ int tag_to_send;
+
+ int timeout;
+ int conn_attempts;
+ time_t last_attempt;
+
+ Timer *timer;
+
+ /* the configured audio format */
+ struct audio_format audio_format;
+
+ struct shout_buffer buf;
+};
+
+extern const struct shout_encoder_plugin shout_mp3_encoder;
+extern const struct shout_encoder_plugin shout_ogg_encoder;
+
+#endif
+
+#endif
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-06 06:44:00 +0000